Browse code
added changes to implement the From number drop-down list, the available numbers and default number fields and the debug logging checkbox, etc.
DoubleBastionAdmin authored on 08/01/2024 19:33:20
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+<span style="display:block;height:15px!important"></span> |
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+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
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+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, or without Asterisk, if it's connected directly to the |
|
| 9 |
+SIP provider. For calls to and from regular phone numbers, a SIP provider like Telnyx or Localphone is needed and a real phone number acquired from that provider. If Asterisk |
|
| 10 |
+is used, it's recommended to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or dedicated server, as explained in the documentation mentioned in the 'Installation' |
|
| 11 |
+section from below. The web server has to be configured to allow access to a specific directory and to proxy WebSocket traffic to a specific URL, as explained in the documentation. |
|
| 12 |
+Not all SIP providers allow connections from external Asterisk servers or direct connections from web applications that use SIP over WebSocket, like SIP Trip Phone. Thus, you can |
|
| 13 |
+connect this application to Telnyx, Localphone, Twilio, Flowroute, Vonage, etc. via an Asterisk server, but if you want to connect it directly to the SIP provider, from the 5 |
|
| 14 |
+mentioned providers, only Telnyx will work, because the others don't allow direct connections from web applications using SIP over WebSocket. SIP Trip Phone is based on the ctxSip |
|
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+phone. |
|
| 16 |
+ |
|
| 17 |
+<span style="display:block;height:20px!important"></span> |
|
| 18 |
+ |
|
| 19 |
+## Features |
|
| 20 |
+<span style="display:block;height:10px!important"></span> |
|
| 21 |
+ |
|
| 22 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 23 |
+ |
|
| 24 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 25 |
+ |
|
| 26 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 27 |
+ |
|
| 28 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows holding, muting and transferring phone calls. |
|
| 29 |
+ |
|
| 30 |
+* 📑 When using Asterisk, SIP Trip Phone allows choosing any available phone number as the 'From' number for outgoing calls. |
|
| 31 |
+ |
|
| 32 |
+* 🚩 Incoming calls are signaled by on-screen notifications. |
|
| 33 |
+ |
|
| 34 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 35 |
+ |
|
| 36 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 37 |
+ |
|
| 38 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 39 |
+ |
|
| 40 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 41 |
+ |
|
| 42 |
+Double Bastion is not affiliated with Telnyx, Localphone, Twilio, Flowroute or Vonage. |
|
| 43 |
+ |
|
| 44 |
+### Donations |
|
| 45 |
+ |
|
| 46 |
+* 🎁 [Donate](https://www.doublebastion.com/donations/) |
|
| 47 |
+ |
|
| 48 |
+<span style="display:block;height:20px!important"></span> |
|
| 49 |
+ |
|
| 50 |
+<p align="center">Initial screen</p> |
|
| 51 |
+<span style="display:block;height:10px!important"></span> |
|
| 52 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 53 |
+<span style="display:block;height:40px!important"></span> |
|
| 54 |
+<p align="center">Dialpad</p> |
|
| 55 |
+<span style="display:block;height:10px!important"></span> |
|
| 56 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 57 |
+<span style="display:block;height:40px!important"></span> |
|
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+<p align="center">Making calls</p> |
|
| 59 |
+<span style="display:block;height:10px!important"></span> |
|
| 60 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
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+<span style="display:block;height:40px!important"></span> |
|
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+<p align="center">Transferring calls</p> |
|
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+<span style="display:block;height:10px!important"></span> |
|
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+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 65 |
+<span style="display:block;height:40px!important"></span> |
|
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+ |
|
| 67 |
+## Browsers |
|
| 68 |
+<span style="display:block;height:10px!important"></span> |
|
| 69 |
+ |
|
| 70 |
+SIP Trip Phone works with all the major browsers. |
|
| 71 |
+ |
|
| 72 |
+<span style="display:block;height:20px!important"></span> |
|
| 73 |
+ |
|
| 74 |
+## Programming Languages |
|
| 75 |
+<span style="display:block;height:10px!important"></span> |
|
| 76 |
+ |
|
| 77 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 78 |
+ |
|
| 79 |
+<span style="display:block;height:20px!important"></span> |
|
| 80 |
+ |
|
| 81 |
+## Minimum Requirements |
|
| 82 |
+<span style="display:block;height:10px!important"></span> |
|
| 83 |
+ |
|
| 84 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 85 |
+ |
|
| 86 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from web applications that use SIP over WebSocket. |
|
| 87 |
+ |
|
| 88 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk, preferably version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or |
|
| 89 |
+dedicated server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 90 |
+ |
|
| 91 |
+<span style="display:block;height:20px!important"></span> |
|
| 92 |
+ |
|
| 93 |
+## Installation |
|
| 94 |
+<span style="display:block;height:10px!important"></span> |
|
| 95 |
+ |
|
| 96 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 97 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 98 |
+ |
|
| 99 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 100 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 101 |
+ |
|
| 102 |
+<span style="display:block;height:20px!important"></span> |
|
| 103 |
+ |
|
| 104 |
+## Contribute |
|
| 105 |
+<span style="display:block;height:10px!important"></span> |
|
| 106 |
+ |
|
| 107 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 108 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 109 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 110 |
+ |
|
| 111 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 112 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 113 |
+project. |
|
| 114 |
+ |
|
| 115 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 116 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 117 |
+ |
|
| 118 |
+<span style="display:block;height:20px!important"></span> |
|
| 119 |
+ |
|
| 120 |
+## License |
|
| 121 |
+<span style="display:block;height:10px!important"></span> |
|
| 122 |
+ |
|
| 123 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 124 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 125 |
+ |
|
| 126 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 127 |
+ |
|
| 128 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed appinfo/info.xml appinfo/signature.json CHANGELOG.txt README.md js/settings.js js/launchphone.js css/style.css phone/scripts/app.js templates/settings.php l10n/en_GB.js l10n/en_GB.json phone/css/ctxSip.css phone/index.html lib/Controller/SphoneController.php lib/Service/SphoneService.php phone/sounds/dtmf.mp3 phone/sounds/incoming.mp3 phone/sounds/outgoing.mp3 img/sip_trip_phone_screenshot.png img/sip_trip_phone_transfer_call.png img/sip_trip_phone_making_calls.png img/sip_trip_phone_initial_screen.png img/sip_trip_phone_dialpad.png
DoubleBastionAdmin authored on 08/01/2024 19:00:48
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
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@@ -1,126 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
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-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
- |
|
| 8 |
-It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, or without Asterisk, if connected directly to the |
|
| 9 |
-SIP provider. For calls to and from regular phone numbers, a SIP provider like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk |
|
| 10 |
-is used, it's recommended to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or dedicated server, as explained in the documentation mentioned in the 'Installation' |
|
| 11 |
-section from below. The web server has to be configured to allow access to a specific directory and to proxy WebSocket traffic to a specific URL, as explained in the documentation. |
|
| 12 |
-Not all SIP providers allow connections from external Asterisk servers or direct connections from web applications that use SIP over WebSocket, like SIP Trip Phone. Thus, you can |
|
| 13 |
-connect this application to Telnyx, Localphone, Twilio, Flowroute, Vonage, etc. via an Asterisk server, but if you want to connect it directly to the SIP provider, from the 5 |
|
| 14 |
-mentioned providers, only Telnyx will work, because the others don't allow direct connections from web applications using SIP over WebSocket. SIP Trip Phone is based on the ctxSip |
|
| 15 |
-phone. |
|
| 16 |
- |
|
| 17 |
-<span style="display:block;height:20px!important"></span> |
|
| 18 |
- |
|
| 19 |
-## Features |
|
| 20 |
-<span style="display:block;height:10px!important"></span> |
|
| 21 |
- |
|
| 22 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 23 |
- |
|
| 24 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 25 |
- |
|
| 26 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 27 |
- |
|
| 28 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 29 |
- |
|
| 30 |
-* 🚩 Incoming calls are signaled by on-screen notifications. |
|
| 31 |
- |
|
| 32 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 33 |
- |
|
| 34 |
-* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 35 |
- |
|
| 36 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 37 |
- |
|
| 38 |
-* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 39 |
- |
|
| 40 |
-Double Bastion is not affiliated with Telnyx, Localphone, Twilio, Flowroute or Vonage. |
|
| 41 |
- |
|
| 42 |
-### Donations |
|
| 43 |
- |
|
| 44 |
-* 🎁 [Donate](https://www.doublebastion.com/donations/) |
|
| 45 |
- |
|
| 46 |
-<span style="display:block;height:20px!important"></span> |
|
| 47 |
- |
|
| 48 |
-<p align="center">Initial screen</p> |
|
| 49 |
-<span style="display:block;height:10px!important"></span> |
|
| 50 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 51 |
-<span style="display:block;height:40px!important"></span> |
|
| 52 |
-<p align="center">Dialpad</p> |
|
| 53 |
-<span style="display:block;height:10px!important"></span> |
|
| 54 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 55 |
-<span style="display:block;height:40px!important"></span> |
|
| 56 |
-<p align="center">Making calls</p> |
|
| 57 |
-<span style="display:block;height:10px!important"></span> |
|
| 58 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 59 |
-<span style="display:block;height:40px!important"></span> |
|
| 60 |
-<p align="center">Transferring calls</p> |
|
| 61 |
-<span style="display:block;height:10px!important"></span> |
|
| 62 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 63 |
-<span style="display:block;height:40px!important"></span> |
|
| 64 |
- |
|
| 65 |
-## Browsers |
|
| 66 |
-<span style="display:block;height:10px!important"></span> |
|
| 67 |
- |
|
| 68 |
-SIP Trip Phone works with all the major browsers. |
|
| 69 |
- |
|
| 70 |
-<span style="display:block;height:20px!important"></span> |
|
| 71 |
- |
|
| 72 |
-## Programming Languages |
|
| 73 |
-<span style="display:block;height:10px!important"></span> |
|
| 74 |
- |
|
| 75 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 76 |
- |
|
| 77 |
-<span style="display:block;height:20px!important"></span> |
|
| 78 |
- |
|
| 79 |
-## Minimum Requirements |
|
| 80 |
-<span style="display:block;height:10px!important"></span> |
|
| 81 |
- |
|
| 82 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 83 |
- |
|
| 84 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from web applications that use SIP over WebSocket. |
|
| 85 |
- |
|
| 86 |
-If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk, preferably version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or |
|
| 87 |
-dedicated server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 88 |
- |
|
| 89 |
-<span style="display:block;height:20px!important"></span> |
|
| 90 |
- |
|
| 91 |
-## Installation |
|
| 92 |
-<span style="display:block;height:10px!important"></span> |
|
| 93 |
- |
|
| 94 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 95 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 96 |
- |
|
| 97 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 98 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 99 |
- |
|
| 100 |
-<span style="display:block;height:20px!important"></span> |
|
| 101 |
- |
|
| 102 |
-## Contribute |
|
| 103 |
-<span style="display:block;height:10px!important"></span> |
|
| 104 |
- |
|
| 105 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 106 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 107 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 108 |
- |
|
| 109 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 110 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 111 |
-project. |
|
| 112 |
- |
|
| 113 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 114 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 115 |
- |
|
| 116 |
-<span style="display:block;height:20px!important"></span> |
|
| 117 |
- |
|
| 118 |
-## License |
|
| 119 |
-<span style="display:block;height:10px!important"></span> |
|
| 120 |
- |
|
| 121 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 122 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 123 |
- |
|
| 124 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 125 |
- |
|
| 126 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added appinfo/info.xml appinfo/signature.json Contributors.txt CHANGELOG.txt README.md lib/Settings/Personal.php css/style.css js/launchphone.js
DoubleBastionAdmin authored on 27/10/2022 16:22:02
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,126 @@ |
| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, or without Asterisk, if connected directly to the |
|
| 9 |
+SIP provider. For calls to and from regular phone numbers, a SIP provider like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk |
|
| 10 |
+is used, it's recommended to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or dedicated server, as explained in the documentation mentioned in the 'Installation' |
|
| 11 |
+section from below. The web server has to be configured to allow access to a specific directory and to proxy WebSocket traffic to a specific URL, as explained in the documentation. |
|
| 12 |
+Not all SIP providers allow connections from external Asterisk servers or direct connections from web applications that use SIP over WebSocket, like SIP Trip Phone. Thus, you can |
|
| 13 |
+connect this application to Telnyx, Localphone, Twilio, Flowroute, Vonage, etc. via an Asterisk server, but if you want to connect it directly to the SIP provider, from the 5 |
|
| 14 |
+mentioned providers, only Telnyx will work, because the others don't allow direct connections from web applications using SIP over WebSocket. SIP Trip Phone is based on the ctxSip |
|
| 15 |
+phone. |
|
| 16 |
+ |
|
| 17 |
+<span style="display:block;height:20px!important"></span> |
|
| 18 |
+ |
|
| 19 |
+## Features |
|
| 20 |
+<span style="display:block;height:10px!important"></span> |
|
| 21 |
+ |
|
| 22 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 23 |
+ |
|
| 24 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 25 |
+ |
|
| 26 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 27 |
+ |
|
| 28 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 29 |
+ |
|
| 30 |
+* 🚩 Incoming calls are signaled by on-screen notifications. |
|
| 31 |
+ |
|
| 32 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 33 |
+ |
|
| 34 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 35 |
+ |
|
| 36 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 37 |
+ |
|
| 38 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 39 |
+ |
|
| 40 |
+Double Bastion is not affiliated with Telnyx, Localphone, Twilio, Flowroute or Vonage. |
|
| 41 |
+ |
|
| 42 |
+### Donations |
|
| 43 |
+ |
|
| 44 |
+* 🎁 [Donate](https://www.doublebastion.com/donations/) |
|
| 45 |
+ |
|
| 46 |
+<span style="display:block;height:20px!important"></span> |
|
| 47 |
+ |
|
| 48 |
+<p align="center">Initial screen</p> |
|
| 49 |
+<span style="display:block;height:10px!important"></span> |
|
| 50 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 51 |
+<span style="display:block;height:40px!important"></span> |
|
| 52 |
+<p align="center">Dialpad</p> |
|
| 53 |
+<span style="display:block;height:10px!important"></span> |
|
| 54 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 55 |
+<span style="display:block;height:40px!important"></span> |
|
| 56 |
+<p align="center">Making calls</p> |
|
| 57 |
+<span style="display:block;height:10px!important"></span> |
|
| 58 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 59 |
+<span style="display:block;height:40px!important"></span> |
|
| 60 |
+<p align="center">Transferring calls</p> |
|
| 61 |
+<span style="display:block;height:10px!important"></span> |
|
| 62 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 63 |
+<span style="display:block;height:40px!important"></span> |
|
| 64 |
+ |
|
| 65 |
+## Browsers |
|
| 66 |
+<span style="display:block;height:10px!important"></span> |
|
| 67 |
+ |
|
| 68 |
+SIP Trip Phone works with all the major browsers. |
|
| 69 |
+ |
|
| 70 |
+<span style="display:block;height:20px!important"></span> |
|
| 71 |
+ |
|
| 72 |
+## Programming Languages |
|
| 73 |
+<span style="display:block;height:10px!important"></span> |
|
| 74 |
+ |
|
| 75 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 76 |
+ |
|
| 77 |
+<span style="display:block;height:20px!important"></span> |
|
| 78 |
+ |
|
| 79 |
+## Minimum Requirements |
|
| 80 |
+<span style="display:block;height:10px!important"></span> |
|
| 81 |
+ |
|
| 82 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 83 |
+ |
|
| 84 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from web applications that use SIP over WebSocket. |
|
| 85 |
+ |
|
| 86 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk, preferably version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or |
|
| 87 |
+dedicated server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 88 |
+ |
|
| 89 |
+<span style="display:block;height:20px!important"></span> |
|
| 90 |
+ |
|
| 91 |
+## Installation |
|
| 92 |
+<span style="display:block;height:10px!important"></span> |
|
| 93 |
+ |
|
| 94 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 95 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 96 |
+ |
|
| 97 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 98 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 99 |
+ |
|
| 100 |
+<span style="display:block;height:20px!important"></span> |
|
| 101 |
+ |
|
| 102 |
+## Contribute |
|
| 103 |
+<span style="display:block;height:10px!important"></span> |
|
| 104 |
+ |
|
| 105 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 106 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 107 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 108 |
+ |
|
| 109 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 110 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 111 |
+project. |
|
| 112 |
+ |
|
| 113 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 114 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 115 |
+ |
|
| 116 |
+<span style="display:block;height:20px!important"></span> |
|
| 117 |
+ |
|
| 118 |
+## License |
|
| 119 |
+<span style="display:block;height:10px!important"></span> |
|
| 120 |
+ |
|
| 121 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 122 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 123 |
+ |
|
| 124 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 125 |
+ |
|
| 126 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed appinfo/info.xml appinfo/signature.json Contributors.txt CHANGELOG.txt README.md lib/Settings/Personal.php css/style.css js/launchphone.js
DoubleBastionAdmin authored on 27/10/2022 16:18:04
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
| ... | ... |
@@ -1,126 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
- |
|
| 8 |
-It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, or without Asterisk, if connected directly to the |
|
| 9 |
-SIP provider. For calls to and from regular phone numbers, a SIP provider like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk |
|
| 10 |
-is used, it's recommended to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or dedicated server, as explained in the documentation mentioned in the 'Installation' |
|
| 11 |
-section from below. The web server has to be configured to allow access to a specific directory and to proxy WebSocket traffic to a specific URL, as explained in the documentation. |
|
| 12 |
-Not all SIP providers allow connections from external Asterisk servers or direct connections from web applications that use SIP over WebSocket, like SIP Trip Phone. Thus, you can |
|
| 13 |
-connect this application to Telnyx, Localphone, Twilio, Flowroute, Vonage, etc. via an Asterisk server, but if you want to connect it directly to the SIP provider, from the 5 |
|
| 14 |
-mentioned providers, only Telnyx will work, because the others don't allow direct connections from web applications using SIP over WebSocket. SIP Trip Phone is based on the ctxSip |
|
| 15 |
-phone. |
|
| 16 |
- |
|
| 17 |
-<span style="display:block;height:20px!important"></span> |
|
| 18 |
- |
|
| 19 |
-## Features |
|
| 20 |
-<span style="display:block;height:10px!important"></span> |
|
| 21 |
- |
|
| 22 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 23 |
- |
|
| 24 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 25 |
- |
|
| 26 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 27 |
- |
|
| 28 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 29 |
- |
|
| 30 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 31 |
- |
|
| 32 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 33 |
- |
|
| 34 |
-* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 35 |
- |
|
| 36 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 37 |
- |
|
| 38 |
-* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 39 |
- |
|
| 40 |
-Double Bastion is not affiliated with Telnyx, Localphone, Twilio, Flowroute, Vonage, or with any other SIP provider. |
|
| 41 |
- |
|
| 42 |
-### Donations |
|
| 43 |
- |
|
| 44 |
-* 🎁 [Donate](https://www.doublebastion.com/donations/) |
|
| 45 |
- |
|
| 46 |
-<span style="display:block;height:20px!important"></span> |
|
| 47 |
- |
|
| 48 |
-<p align="center">Initial screen</p> |
|
| 49 |
-<span style="display:block;height:10px!important"></span> |
|
| 50 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 51 |
-<span style="display:block;height:40px!important"></span> |
|
| 52 |
-<p align="center">Dialpad</p> |
|
| 53 |
-<span style="display:block;height:10px!important"></span> |
|
| 54 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 55 |
-<span style="display:block;height:40px!important"></span> |
|
| 56 |
-<p align="center">Making calls</p> |
|
| 57 |
-<span style="display:block;height:10px!important"></span> |
|
| 58 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 59 |
-<span style="display:block;height:40px!important"></span> |
|
| 60 |
-<p align="center">Transferring calls</p> |
|
| 61 |
-<span style="display:block;height:10px!important"></span> |
|
| 62 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 63 |
-<span style="display:block;height:40px!important"></span> |
|
| 64 |
- |
|
| 65 |
-## Browsers |
|
| 66 |
-<span style="display:block;height:10px!important"></span> |
|
| 67 |
- |
|
| 68 |
-SIP Trip Phone works with all the major browsers. |
|
| 69 |
- |
|
| 70 |
-<span style="display:block;height:20px!important"></span> |
|
| 71 |
- |
|
| 72 |
-## Programming Languages |
|
| 73 |
-<span style="display:block;height:10px!important"></span> |
|
| 74 |
- |
|
| 75 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 76 |
- |
|
| 77 |
-<span style="display:block;height:20px!important"></span> |
|
| 78 |
- |
|
| 79 |
-## Minimum Requirements |
|
| 80 |
-<span style="display:block;height:10px!important"></span> |
|
| 81 |
- |
|
| 82 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 83 |
- |
|
| 84 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from web applications that use SIP over WebSocket. |
|
| 85 |
- |
|
| 86 |
-If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk, preferably version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or |
|
| 87 |
-dedicated server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 88 |
- |
|
| 89 |
-<span style="display:block;height:20px!important"></span> |
|
| 90 |
- |
|
| 91 |
-## Installation |
|
| 92 |
-<span style="display:block;height:10px!important"></span> |
|
| 93 |
- |
|
| 94 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 95 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 96 |
- |
|
| 97 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 98 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 99 |
- |
|
| 100 |
-<span style="display:block;height:20px!important"></span> |
|
| 101 |
- |
|
| 102 |
-## Contribute |
|
| 103 |
-<span style="display:block;height:10px!important"></span> |
|
| 104 |
- |
|
| 105 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 106 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 107 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 108 |
- |
|
| 109 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 110 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 111 |
-project. |
|
| 112 |
- |
|
| 113 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 114 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 115 |
- |
|
| 116 |
-<span style="display:block;height:20px!important"></span> |
|
| 117 |
- |
|
| 118 |
-## License |
|
| 119 |
-<span style="display:block;height:10px!important"></span> |
|
| 120 |
- |
|
| 121 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 122 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 123 |
- |
|
| 124 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 125 |
- |
|
| 126 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added appinfo/signature.json README.md
DoubleBastionAdmin authored on 22/09/2022 09:05:02
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,126 @@ |
| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, or without Asterisk, if connected directly to the |
|
| 9 |
+SIP provider. For calls to and from regular phone numbers, a SIP provider like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk |
|
| 10 |
+is used, it's recommended to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or dedicated server, as explained in the documentation mentioned in the 'Installation' |
|
| 11 |
+section from below. The web server has to be configured to allow access to a specific directory and to proxy WebSocket traffic to a specific URL, as explained in the documentation. |
|
| 12 |
+Not all SIP providers allow connections from external Asterisk servers or direct connections from web applications that use SIP over WebSocket, like SIP Trip Phone. Thus, you can |
|
| 13 |
+connect this application to Telnyx, Localphone, Twilio, Flowroute, Vonage, etc. via an Asterisk server, but if you want to connect it directly to the SIP provider, from the 5 |
|
| 14 |
+mentioned providers, only Telnyx will work, because the others don't allow direct connections from web applications using SIP over WebSocket. SIP Trip Phone is based on the ctxSip |
|
| 15 |
+phone. |
|
| 16 |
+ |
|
| 17 |
+<span style="display:block;height:20px!important"></span> |
|
| 18 |
+ |
|
| 19 |
+## Features |
|
| 20 |
+<span style="display:block;height:10px!important"></span> |
|
| 21 |
+ |
|
| 22 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 23 |
+ |
|
| 24 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 25 |
+ |
|
| 26 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 27 |
+ |
|
| 28 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 29 |
+ |
|
| 30 |
+* 🚩 On-screen notifications on incoming calls. |
|
| 31 |
+ |
|
| 32 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 33 |
+ |
|
| 34 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 35 |
+ |
|
| 36 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 37 |
+ |
|
| 38 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 39 |
+ |
|
| 40 |
+Double Bastion is not affiliated with Telnyx, Localphone, Twilio, Flowroute, Vonage, or with any other SIP provider. |
|
| 41 |
+ |
|
| 42 |
+### Donations |
|
| 43 |
+ |
|
| 44 |
+* 🎁 [Donate](https://www.doublebastion.com/donations/) |
|
| 45 |
+ |
|
| 46 |
+<span style="display:block;height:20px!important"></span> |
|
| 47 |
+ |
|
| 48 |
+<p align="center">Initial screen</p> |
|
| 49 |
+<span style="display:block;height:10px!important"></span> |
|
| 50 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 51 |
+<span style="display:block;height:40px!important"></span> |
|
| 52 |
+<p align="center">Dialpad</p> |
|
| 53 |
+<span style="display:block;height:10px!important"></span> |
|
| 54 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 55 |
+<span style="display:block;height:40px!important"></span> |
|
| 56 |
+<p align="center">Making calls</p> |
|
| 57 |
+<span style="display:block;height:10px!important"></span> |
|
| 58 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 59 |
+<span style="display:block;height:40px!important"></span> |
|
| 60 |
+<p align="center">Transferring calls</p> |
|
| 61 |
+<span style="display:block;height:10px!important"></span> |
|
| 62 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 63 |
+<span style="display:block;height:40px!important"></span> |
|
| 64 |
+ |
|
| 65 |
+## Browsers |
|
| 66 |
+<span style="display:block;height:10px!important"></span> |
|
| 67 |
+ |
|
| 68 |
+SIP Trip Phone works with all the major browsers. |
|
| 69 |
+ |
|
| 70 |
+<span style="display:block;height:20px!important"></span> |
|
| 71 |
+ |
|
| 72 |
+## Programming Languages |
|
| 73 |
+<span style="display:block;height:10px!important"></span> |
|
| 74 |
+ |
|
| 75 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 76 |
+ |
|
| 77 |
+<span style="display:block;height:20px!important"></span> |
|
| 78 |
+ |
|
| 79 |
+## Minimum Requirements |
|
| 80 |
+<span style="display:block;height:10px!important"></span> |
|
| 81 |
+ |
|
| 82 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 83 |
+ |
|
| 84 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from web applications that use SIP over WebSocket. |
|
| 85 |
+ |
|
| 86 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk, preferably version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or |
|
| 87 |
+dedicated server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 88 |
+ |
|
| 89 |
+<span style="display:block;height:20px!important"></span> |
|
| 90 |
+ |
|
| 91 |
+## Installation |
|
| 92 |
+<span style="display:block;height:10px!important"></span> |
|
| 93 |
+ |
|
| 94 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 95 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 96 |
+ |
|
| 97 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 98 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 99 |
+ |
|
| 100 |
+<span style="display:block;height:20px!important"></span> |
|
| 101 |
+ |
|
| 102 |
+## Contribute |
|
| 103 |
+<span style="display:block;height:10px!important"></span> |
|
| 104 |
+ |
|
| 105 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 106 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 107 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 108 |
+ |
|
| 109 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 110 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 111 |
+project. |
|
| 112 |
+ |
|
| 113 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 114 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 115 |
+ |
|
| 116 |
+<span style="display:block;height:20px!important"></span> |
|
| 117 |
+ |
|
| 118 |
+## License |
|
| 119 |
+<span style="display:block;height:10px!important"></span> |
|
| 120 |
+ |
|
| 121 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 122 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 123 |
+ |
|
| 124 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 125 |
+ |
|
| 126 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed appinfo/signature.json README.md
DoubleBastionAdmin authored on 22/09/2022 09:03:13
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
| ... | ... |
@@ -1,126 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
- |
|
| 8 |
-It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, or without Asterisk, if connected directly to the |
|
| 9 |
-SIP provider. For calls to and from regular phone numbers, a SIP provider like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk |
|
| 10 |
-is used, it's recommended to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or dedicated server, as explained in the documentation mentioned in the 'Installation' |
|
| 11 |
-section from below. The web server has to be configured to allow access to a specific directory and to proxy WebSocket traffic to a specific URL, as explained in the documentation. |
|
| 12 |
-Not all SIP providers allow connections from external Asterisk servers or direct connections from web applications that use SIP over WebSocket, like SIP Trip Phone. Thus, you can |
|
| 13 |
-connect this application to Telnyx, Localphone, Twilio, Flowroute, Vonage, etc. via an Asterisk server, but if you want to connect it directly to the SIP provider, from the 5 |
|
| 14 |
-mentioned providers, only Telnyx will work, because the others don't allow direct connections from web applications using SIP over WebSocket. SIP Trip Phone is based on the ctxSip |
|
| 15 |
-phone. |
|
| 16 |
- |
|
| 17 |
-<span style="display:block;height:20px!important"></span> |
|
| 18 |
- |
|
| 19 |
-## Features |
|
| 20 |
-<span style="display:block;height:10px!important"></span> |
|
| 21 |
- |
|
| 22 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 23 |
- |
|
| 24 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 25 |
- |
|
| 26 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 27 |
- |
|
| 28 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 29 |
- |
|
| 30 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 31 |
- |
|
| 32 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 33 |
- |
|
| 34 |
-* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 35 |
- |
|
| 36 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 37 |
- |
|
| 38 |
-* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 39 |
- |
|
| 40 |
-Double Bastion is not affiliated with Telnyx, Localphone, Twilio, Flowroute, Vonage, or with any other SIP provider. |
|
| 41 |
- |
|
| 42 |
-### Donations |
|
| 43 |
- |
|
| 44 |
-* [Donations are welcome](https://www.doublebastion.com/donations/) |
|
| 45 |
- |
|
| 46 |
-<span style="display:block;height:20px!important"></span> |
|
| 47 |
- |
|
| 48 |
-<p align="center">Initial screen</p> |
|
| 49 |
-<span style="display:block;height:10px!important"></span> |
|
| 50 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 51 |
-<span style="display:block;height:40px!important"></span> |
|
| 52 |
-<p align="center">Dialpad</p> |
|
| 53 |
-<span style="display:block;height:10px!important"></span> |
|
| 54 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 55 |
-<span style="display:block;height:40px!important"></span> |
|
| 56 |
-<p align="center">Making calls</p> |
|
| 57 |
-<span style="display:block;height:10px!important"></span> |
|
| 58 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 59 |
-<span style="display:block;height:40px!important"></span> |
|
| 60 |
-<p align="center">Transferring calls</p> |
|
| 61 |
-<span style="display:block;height:10px!important"></span> |
|
| 62 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 63 |
-<span style="display:block;height:40px!important"></span> |
|
| 64 |
- |
|
| 65 |
-## Browsers |
|
| 66 |
-<span style="display:block;height:10px!important"></span> |
|
| 67 |
- |
|
| 68 |
-SIP Trip Phone works with all the major browsers. |
|
| 69 |
- |
|
| 70 |
-<span style="display:block;height:20px!important"></span> |
|
| 71 |
- |
|
| 72 |
-## Programming Languages |
|
| 73 |
-<span style="display:block;height:10px!important"></span> |
|
| 74 |
- |
|
| 75 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 76 |
- |
|
| 77 |
-<span style="display:block;height:20px!important"></span> |
|
| 78 |
- |
|
| 79 |
-## Minimum Requirements |
|
| 80 |
-<span style="display:block;height:10px!important"></span> |
|
| 81 |
- |
|
| 82 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 83 |
- |
|
| 84 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from web applications that use SIP over WebSocket. |
|
| 85 |
- |
|
| 86 |
-If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk, preferably version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or |
|
| 87 |
-dedicated server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 88 |
- |
|
| 89 |
-<span style="display:block;height:20px!important"></span> |
|
| 90 |
- |
|
| 91 |
-## Installation |
|
| 92 |
-<span style="display:block;height:10px!important"></span> |
|
| 93 |
- |
|
| 94 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 95 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 96 |
- |
|
| 97 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 98 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 99 |
- |
|
| 100 |
-<span style="display:block;height:20px!important"></span> |
|
| 101 |
- |
|
| 102 |
-## Contribute |
|
| 103 |
-<span style="display:block;height:10px!important"></span> |
|
| 104 |
- |
|
| 105 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 106 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 107 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 108 |
- |
|
| 109 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 110 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 111 |
-project. |
|
| 112 |
- |
|
| 113 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 114 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 115 |
- |
|
| 116 |
-<span style="display:block;height:20px!important"></span> |
|
| 117 |
- |
|
| 118 |
-## License |
|
| 119 |
-<span style="display:block;height:10px!important"></span> |
|
| 120 |
- |
|
| 121 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 122 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 123 |
- |
|
| 124 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 125 |
- |
|
| 126 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added CHANGELOG.txt README.md appinfo/info.xml appinfo/signature.json phone/scripts/app.js templates/settings.php css/style.css
DoubleBastionAdmin authored on 22/09/2022 08:46:22
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,126 @@ |
| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, or without Asterisk, if connected directly to the |
|
| 9 |
+SIP provider. For calls to and from regular phone numbers, a SIP provider like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk |
|
| 10 |
+is used, it's recommended to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or dedicated server, as explained in the documentation mentioned in the 'Installation' |
|
| 11 |
+section from below. The web server has to be configured to allow access to a specific directory and to proxy WebSocket traffic to a specific URL, as explained in the documentation. |
|
| 12 |
+Not all SIP providers allow connections from external Asterisk servers or direct connections from web applications that use SIP over WebSocket, like SIP Trip Phone. Thus, you can |
|
| 13 |
+connect this application to Telnyx, Localphone, Twilio, Flowroute, Vonage, etc. via an Asterisk server, but if you want to connect it directly to the SIP provider, from the 5 |
|
| 14 |
+mentioned providers, only Telnyx will work, because the others don't allow direct connections from web applications using SIP over WebSocket. SIP Trip Phone is based on the ctxSip |
|
| 15 |
+phone. |
|
| 16 |
+ |
|
| 17 |
+<span style="display:block;height:20px!important"></span> |
|
| 18 |
+ |
|
| 19 |
+## Features |
|
| 20 |
+<span style="display:block;height:10px!important"></span> |
|
| 21 |
+ |
|
| 22 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 23 |
+ |
|
| 24 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 25 |
+ |
|
| 26 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 27 |
+ |
|
| 28 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 29 |
+ |
|
| 30 |
+* 🚩 On-screen notifications on incoming calls. |
|
| 31 |
+ |
|
| 32 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 33 |
+ |
|
| 34 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 35 |
+ |
|
| 36 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 37 |
+ |
|
| 38 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 39 |
+ |
|
| 40 |
+Double Bastion is not affiliated with Telnyx, Localphone, Twilio, Flowroute, Vonage, or with any other SIP provider. |
|
| 41 |
+ |
|
| 42 |
+### Donations |
|
| 43 |
+ |
|
| 44 |
+* [Donations are welcome](https://www.doublebastion.com/donations/) |
|
| 45 |
+ |
|
| 46 |
+<span style="display:block;height:20px!important"></span> |
|
| 47 |
+ |
|
| 48 |
+<p align="center">Initial screen</p> |
|
| 49 |
+<span style="display:block;height:10px!important"></span> |
|
| 50 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 51 |
+<span style="display:block;height:40px!important"></span> |
|
| 52 |
+<p align="center">Dialpad</p> |
|
| 53 |
+<span style="display:block;height:10px!important"></span> |
|
| 54 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 55 |
+<span style="display:block;height:40px!important"></span> |
|
| 56 |
+<p align="center">Making calls</p> |
|
| 57 |
+<span style="display:block;height:10px!important"></span> |
|
| 58 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 59 |
+<span style="display:block;height:40px!important"></span> |
|
| 60 |
+<p align="center">Transferring calls</p> |
|
| 61 |
+<span style="display:block;height:10px!important"></span> |
|
| 62 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 63 |
+<span style="display:block;height:40px!important"></span> |
|
| 64 |
+ |
|
| 65 |
+## Browsers |
|
| 66 |
+<span style="display:block;height:10px!important"></span> |
|
| 67 |
+ |
|
| 68 |
+SIP Trip Phone works with all the major browsers. |
|
| 69 |
+ |
|
| 70 |
+<span style="display:block;height:20px!important"></span> |
|
| 71 |
+ |
|
| 72 |
+## Programming Languages |
|
| 73 |
+<span style="display:block;height:10px!important"></span> |
|
| 74 |
+ |
|
| 75 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 76 |
+ |
|
| 77 |
+<span style="display:block;height:20px!important"></span> |
|
| 78 |
+ |
|
| 79 |
+## Minimum Requirements |
|
| 80 |
+<span style="display:block;height:10px!important"></span> |
|
| 81 |
+ |
|
| 82 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 83 |
+ |
|
| 84 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from web applications that use SIP over WebSocket. |
|
| 85 |
+ |
|
| 86 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk, preferably version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or |
|
| 87 |
+dedicated server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 88 |
+ |
|
| 89 |
+<span style="display:block;height:20px!important"></span> |
|
| 90 |
+ |
|
| 91 |
+## Installation |
|
| 92 |
+<span style="display:block;height:10px!important"></span> |
|
| 93 |
+ |
|
| 94 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 95 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 96 |
+ |
|
| 97 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 98 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 99 |
+ |
|
| 100 |
+<span style="display:block;height:20px!important"></span> |
|
| 101 |
+ |
|
| 102 |
+## Contribute |
|
| 103 |
+<span style="display:block;height:10px!important"></span> |
|
| 104 |
+ |
|
| 105 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 106 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 107 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 108 |
+ |
|
| 109 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 110 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 111 |
+project. |
|
| 112 |
+ |
|
| 113 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 114 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 115 |
+ |
|
| 116 |
+<span style="display:block;height:20px!important"></span> |
|
| 117 |
+ |
|
| 118 |
+## License |
|
| 119 |
+<span style="display:block;height:10px!important"></span> |
|
| 120 |
+ |
|
| 121 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 122 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 123 |
+ |
|
| 124 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 125 |
+ |
|
| 126 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed CHANGELOG.txt README.md appinfo/info.xml appinfo/signature.json phone/scripts/app.js templates/settings.php css/style.css
DoubleBastionAdmin authored on 22/09/2022 08:32:29
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
| ... | ... |
@@ -1,125 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
- |
|
| 8 |
-It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, but also without Asterisk, if you connect it directly |
|
| 9 |
-to the SIP provider (this implies that the SIP provider allows direct connections from WebRTC applications). For calls to and from regular phone numbers, a SIP provider like Telnyx or |
|
| 10 |
-Localphone is needed and a real phone number acquired from that SIP provider. SIP Trip Phone will start only if it is properly configured and connected to Asterisk or directly to the |
|
| 11 |
-SIP provider. If Asterisk is used, it's recommended to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or dedicated server, as explained in the the documentation |
|
| 12 |
-mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory and to proxy WebSocket traffic to a specific URL, as |
|
| 13 |
-explained in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from WebRTC applications. Telnyx and Localphone allow direct |
|
| 14 |
-connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip Phone is based on the ctxSip phone. |
|
| 15 |
- |
|
| 16 |
-<span style="display:block;height:20px!important"></span> |
|
| 17 |
- |
|
| 18 |
-## Features |
|
| 19 |
-<span style="display:block;height:10px!important"></span> |
|
| 20 |
- |
|
| 21 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
- |
|
| 23 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
- |
|
| 25 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
- |
|
| 27 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
- |
|
| 29 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
- |
|
| 31 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
- |
|
| 33 |
-* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
- |
|
| 35 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
- |
|
| 37 |
-* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
- |
|
| 39 |
-Double Bastion is not affiliated with Telnyx or Localphone. |
|
| 40 |
- |
|
| 41 |
-### Donations |
|
| 42 |
- |
|
| 43 |
-* [Donations are welcome](https://www.doublebastion.com/donations/) |
|
| 44 |
- |
|
| 45 |
-<span style="display:block;height:20px!important"></span> |
|
| 46 |
- |
|
| 47 |
-<p align="center">Initial screen</p> |
|
| 48 |
-<span style="display:block;height:10px!important"></span> |
|
| 49 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
-<span style="display:block;height:40px!important"></span> |
|
| 51 |
-<p align="center">Dialpad</p> |
|
| 52 |
-<span style="display:block;height:10px!important"></span> |
|
| 53 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
-<span style="display:block;height:40px!important"></span> |
|
| 55 |
-<p align="center">Making calls</p> |
|
| 56 |
-<span style="display:block;height:10px!important"></span> |
|
| 57 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
-<span style="display:block;height:40px!important"></span> |
|
| 59 |
-<p align="center">Transferring calls</p> |
|
| 60 |
-<span style="display:block;height:10px!important"></span> |
|
| 61 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 62 |
-<span style="display:block;height:40px!important"></span> |
|
| 63 |
- |
|
| 64 |
-## Browsers |
|
| 65 |
-<span style="display:block;height:10px!important"></span> |
|
| 66 |
- |
|
| 67 |
-SIP Trip Phone works with all the major browsers. |
|
| 68 |
- |
|
| 69 |
-<span style="display:block;height:20px!important"></span> |
|
| 70 |
- |
|
| 71 |
-## Programming Languages |
|
| 72 |
-<span style="display:block;height:10px!important"></span> |
|
| 73 |
- |
|
| 74 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 75 |
- |
|
| 76 |
-<span style="display:block;height:20px!important"></span> |
|
| 77 |
- |
|
| 78 |
-## Minimum Requirements |
|
| 79 |
-<span style="display:block;height:10px!important"></span> |
|
| 80 |
- |
|
| 81 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 82 |
- |
|
| 83 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 84 |
- |
|
| 85 |
-If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk, preferably version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or |
|
| 86 |
-dedicated server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 87 |
- |
|
| 88 |
-<span style="display:block;height:20px!important"></span> |
|
| 89 |
- |
|
| 90 |
-## Installation |
|
| 91 |
-<span style="display:block;height:10px!important"></span> |
|
| 92 |
- |
|
| 93 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 94 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 95 |
- |
|
| 96 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 97 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 98 |
- |
|
| 99 |
-<span style="display:block;height:20px!important"></span> |
|
| 100 |
- |
|
| 101 |
-## Contribute |
|
| 102 |
-<span style="display:block;height:10px!important"></span> |
|
| 103 |
- |
|
| 104 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 105 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 106 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 107 |
- |
|
| 108 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 109 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 110 |
-project. |
|
| 111 |
- |
|
| 112 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 113 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 114 |
- |
|
| 115 |
-<span style="display:block;height:20px!important"></span> |
|
| 116 |
- |
|
| 117 |
-## License |
|
| 118 |
-<span style="display:block;height:10px!important"></span> |
|
| 119 |
- |
|
| 120 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 121 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 122 |
- |
|
| 123 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 124 |
- |
|
| 125 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added appinfo/info.xml appinfo/signature.json CHANGELOG.txt README.md
DoubleBastionAdmin authored on 20/08/2022 11:22:57
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,125 @@ |
| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, but also without Asterisk, if you connect it directly |
|
| 9 |
+to the SIP provider (this implies that the SIP provider allows direct connections from WebRTC applications). For calls to and from regular phone numbers, a SIP provider like Telnyx or |
|
| 10 |
+Localphone is needed and a real phone number acquired from that SIP provider. SIP Trip Phone will start only if it is properly configured and connected to Asterisk or directly to the |
|
| 11 |
+SIP provider. If Asterisk is used, it's recommended to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or dedicated server, as explained in the the documentation |
|
| 12 |
+mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory and to proxy WebSocket traffic to a specific URL, as |
|
| 13 |
+explained in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from WebRTC applications. Telnyx and Localphone allow direct |
|
| 14 |
+connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip Phone is based on the ctxSip phone. |
|
| 15 |
+ |
|
| 16 |
+<span style="display:block;height:20px!important"></span> |
|
| 17 |
+ |
|
| 18 |
+## Features |
|
| 19 |
+<span style="display:block;height:10px!important"></span> |
|
| 20 |
+ |
|
| 21 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
+ |
|
| 23 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
+ |
|
| 25 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
+ |
|
| 27 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
+ |
|
| 29 |
+* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
+ |
|
| 31 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
+ |
|
| 33 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
+ |
|
| 35 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
+ |
|
| 37 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
+ |
|
| 39 |
+Double Bastion is not affiliated with Telnyx or Localphone. |
|
| 40 |
+ |
|
| 41 |
+### Donations |
|
| 42 |
+ |
|
| 43 |
+* [Donations are welcome](https://www.doublebastion.com/donations/) |
|
| 44 |
+ |
|
| 45 |
+<span style="display:block;height:20px!important"></span> |
|
| 46 |
+ |
|
| 47 |
+<p align="center">Initial screen</p> |
|
| 48 |
+<span style="display:block;height:10px!important"></span> |
|
| 49 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
+<span style="display:block;height:40px!important"></span> |
|
| 51 |
+<p align="center">Dialpad</p> |
|
| 52 |
+<span style="display:block;height:10px!important"></span> |
|
| 53 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
+<span style="display:block;height:40px!important"></span> |
|
| 55 |
+<p align="center">Making calls</p> |
|
| 56 |
+<span style="display:block;height:10px!important"></span> |
|
| 57 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
+<span style="display:block;height:40px!important"></span> |
|
| 59 |
+<p align="center">Transferring calls</p> |
|
| 60 |
+<span style="display:block;height:10px!important"></span> |
|
| 61 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 62 |
+<span style="display:block;height:40px!important"></span> |
|
| 63 |
+ |
|
| 64 |
+## Browsers |
|
| 65 |
+<span style="display:block;height:10px!important"></span> |
|
| 66 |
+ |
|
| 67 |
+SIP Trip Phone works with all the major browsers. |
|
| 68 |
+ |
|
| 69 |
+<span style="display:block;height:20px!important"></span> |
|
| 70 |
+ |
|
| 71 |
+## Programming Languages |
|
| 72 |
+<span style="display:block;height:10px!important"></span> |
|
| 73 |
+ |
|
| 74 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 75 |
+ |
|
| 76 |
+<span style="display:block;height:20px!important"></span> |
|
| 77 |
+ |
|
| 78 |
+## Minimum Requirements |
|
| 79 |
+<span style="display:block;height:10px!important"></span> |
|
| 80 |
+ |
|
| 81 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 82 |
+ |
|
| 83 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 84 |
+ |
|
| 85 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk, preferably version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or |
|
| 86 |
+dedicated server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 87 |
+ |
|
| 88 |
+<span style="display:block;height:20px!important"></span> |
|
| 89 |
+ |
|
| 90 |
+## Installation |
|
| 91 |
+<span style="display:block;height:10px!important"></span> |
|
| 92 |
+ |
|
| 93 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 94 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 95 |
+ |
|
| 96 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 97 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 98 |
+ |
|
| 99 |
+<span style="display:block;height:20px!important"></span> |
|
| 100 |
+ |
|
| 101 |
+## Contribute |
|
| 102 |
+<span style="display:block;height:10px!important"></span> |
|
| 103 |
+ |
|
| 104 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 105 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 106 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 107 |
+ |
|
| 108 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 109 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 110 |
+project. |
|
| 111 |
+ |
|
| 112 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 113 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 114 |
+ |
|
| 115 |
+<span style="display:block;height:20px!important"></span> |
|
| 116 |
+ |
|
| 117 |
+## License |
|
| 118 |
+<span style="display:block;height:10px!important"></span> |
|
| 119 |
+ |
|
| 120 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 121 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 122 |
+ |
|
| 123 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 124 |
+ |
|
| 125 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed appinfo/info.xml appinfo/signature.json CHANGELOG.txt README.md
DoubleBastionAdmin authored on 20/08/2022 11:21:13
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
| ... | ... |
@@ -1,121 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
- |
|
| 8 |
-It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, but also without Asterisk, if you connect it directly |
|
| 9 |
-to the SIP provider (this implies that the SIP provider allows direct connections from WebRTC applications). For calls to and from regular phone numbers, a SIP provider like Telnyx or |
|
| 10 |
-Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or |
|
| 11 |
-dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory |
|
| 12 |
-and to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from |
|
| 13 |
-WebRTC applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip |
|
| 14 |
-Phone is based on the ctxSip phone. |
|
| 15 |
- |
|
| 16 |
-<span style="display:block;height:20px!important"></span> |
|
| 17 |
- |
|
| 18 |
-## Main Features |
|
| 19 |
-<span style="display:block;height:10px!important"></span> |
|
| 20 |
- |
|
| 21 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
- |
|
| 23 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
- |
|
| 25 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
- |
|
| 27 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
- |
|
| 29 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
- |
|
| 31 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
- |
|
| 33 |
-* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
- |
|
| 35 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
- |
|
| 37 |
-* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
- |
|
| 39 |
-Double Bastion is not affiliated with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
- |
|
| 41 |
-<span style="display:block;height:20px!important"></span> |
|
| 42 |
- |
|
| 43 |
-<p align="center">Initial screen</p> |
|
| 44 |
-<span style="display:block;height:10px!important"></span> |
|
| 45 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 46 |
-<span style="display:block;height:40px!important"></span> |
|
| 47 |
-<p align="center">Dialpad</p> |
|
| 48 |
-<span style="display:block;height:10px!important"></span> |
|
| 49 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
-<span style="display:block;height:40px!important"></span> |
|
| 51 |
-<p align="center">Making calls</p> |
|
| 52 |
-<span style="display:block;height:10px!important"></span> |
|
| 53 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
-<span style="display:block;height:40px!important"></span> |
|
| 55 |
-<p align="center">Transferring calls</p> |
|
| 56 |
-<span style="display:block;height:10px!important"></span> |
|
| 57 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
-<span style="display:block;height:40px!important"></span> |
|
| 59 |
- |
|
| 60 |
-## Browsers |
|
| 61 |
-<span style="display:block;height:10px!important"></span> |
|
| 62 |
- |
|
| 63 |
-SIP Trip Phone works with all the major browsers. |
|
| 64 |
- |
|
| 65 |
-<span style="display:block;height:20px!important"></span> |
|
| 66 |
- |
|
| 67 |
-## Programming Languages |
|
| 68 |
-<span style="display:block;height:10px!important"></span> |
|
| 69 |
- |
|
| 70 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 71 |
- |
|
| 72 |
-<span style="display:block;height:20px!important"></span> |
|
| 73 |
- |
|
| 74 |
-## Minimum Requirements |
|
| 75 |
-<span style="display:block;height:10px!important"></span> |
|
| 76 |
- |
|
| 77 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 78 |
- |
|
| 79 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 80 |
- |
|
| 81 |
-If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 82 |
-server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 83 |
- |
|
| 84 |
-<span style="display:block;height:20px!important"></span> |
|
| 85 |
- |
|
| 86 |
-## Installation |
|
| 87 |
-<span style="display:block;height:10px!important"></span> |
|
| 88 |
- |
|
| 89 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 90 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 91 |
- |
|
| 92 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 93 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 94 |
- |
|
| 95 |
-<span style="display:block;height:20px!important"></span> |
|
| 96 |
- |
|
| 97 |
-## Contribute |
|
| 98 |
-<span style="display:block;height:10px!important"></span> |
|
| 99 |
- |
|
| 100 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 101 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 102 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 103 |
- |
|
| 104 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 105 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 106 |
-project. |
|
| 107 |
- |
|
| 108 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 109 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 110 |
- |
|
| 111 |
-<span style="display:block;height:20px!important"></span> |
|
| 112 |
- |
|
| 113 |
-## License |
|
| 114 |
-<span style="display:block;height:10px!important"></span> |
|
| 115 |
- |
|
| 116 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 117 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 118 |
- |
|
| 119 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 120 |
- |
|
| 121 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added CHANGELOG.txt Contributors.txt README.md appinfo/info.xml appinfo/signature.json js/launchphone.js lib/Service/SphoneService.php
DoubleBastionAdmin authored on 09/05/2022 18:32:24
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,121 @@ |
| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, but also without Asterisk, if you connect it directly |
|
| 9 |
+to the SIP provider (this implies that the SIP provider allows direct connections from WebRTC applications). For calls to and from regular phone numbers, a SIP provider like Telnyx or |
|
| 10 |
+Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or |
|
| 11 |
+dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory |
|
| 12 |
+and to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from |
|
| 13 |
+WebRTC applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip |
|
| 14 |
+Phone is based on the ctxSip phone. |
|
| 15 |
+ |
|
| 16 |
+<span style="display:block;height:20px!important"></span> |
|
| 17 |
+ |
|
| 18 |
+## Main Features |
|
| 19 |
+<span style="display:block;height:10px!important"></span> |
|
| 20 |
+ |
|
| 21 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
+ |
|
| 23 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
+ |
|
| 25 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
+ |
|
| 27 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
+ |
|
| 29 |
+* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
+ |
|
| 31 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
+ |
|
| 33 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
+ |
|
| 35 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
+ |
|
| 37 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
+ |
|
| 39 |
+Double Bastion is not affiliated with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
+ |
|
| 41 |
+<span style="display:block;height:20px!important"></span> |
|
| 42 |
+ |
|
| 43 |
+<p align="center">Initial screen</p> |
|
| 44 |
+<span style="display:block;height:10px!important"></span> |
|
| 45 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 46 |
+<span style="display:block;height:40px!important"></span> |
|
| 47 |
+<p align="center">Dialpad</p> |
|
| 48 |
+<span style="display:block;height:10px!important"></span> |
|
| 49 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
+<span style="display:block;height:40px!important"></span> |
|
| 51 |
+<p align="center">Making calls</p> |
|
| 52 |
+<span style="display:block;height:10px!important"></span> |
|
| 53 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
+<span style="display:block;height:40px!important"></span> |
|
| 55 |
+<p align="center">Transferring calls</p> |
|
| 56 |
+<span style="display:block;height:10px!important"></span> |
|
| 57 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
+<span style="display:block;height:40px!important"></span> |
|
| 59 |
+ |
|
| 60 |
+## Browsers |
|
| 61 |
+<span style="display:block;height:10px!important"></span> |
|
| 62 |
+ |
|
| 63 |
+SIP Trip Phone works with all the major browsers. |
|
| 64 |
+ |
|
| 65 |
+<span style="display:block;height:20px!important"></span> |
|
| 66 |
+ |
|
| 67 |
+## Programming Languages |
|
| 68 |
+<span style="display:block;height:10px!important"></span> |
|
| 69 |
+ |
|
| 70 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 71 |
+ |
|
| 72 |
+<span style="display:block;height:20px!important"></span> |
|
| 73 |
+ |
|
| 74 |
+## Minimum Requirements |
|
| 75 |
+<span style="display:block;height:10px!important"></span> |
|
| 76 |
+ |
|
| 77 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 78 |
+ |
|
| 79 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 80 |
+ |
|
| 81 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 82 |
+server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 83 |
+ |
|
| 84 |
+<span style="display:block;height:20px!important"></span> |
|
| 85 |
+ |
|
| 86 |
+## Installation |
|
| 87 |
+<span style="display:block;height:10px!important"></span> |
|
| 88 |
+ |
|
| 89 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 90 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 91 |
+ |
|
| 92 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 93 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 94 |
+ |
|
| 95 |
+<span style="display:block;height:20px!important"></span> |
|
| 96 |
+ |
|
| 97 |
+## Contribute |
|
| 98 |
+<span style="display:block;height:10px!important"></span> |
|
| 99 |
+ |
|
| 100 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 101 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 102 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 103 |
+ |
|
| 104 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 105 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 106 |
+project. |
|
| 107 |
+ |
|
| 108 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 109 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 110 |
+ |
|
| 111 |
+<span style="display:block;height:20px!important"></span> |
|
| 112 |
+ |
|
| 113 |
+## License |
|
| 114 |
+<span style="display:block;height:10px!important"></span> |
|
| 115 |
+ |
|
| 116 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 117 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 118 |
+ |
|
| 119 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 120 |
+ |
|
| 121 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed CHANGELOG.txt Contributors.txt README.md appinfo/info.xml appinfo/signature.json js/launchphone.js lib/Service/SphoneService.php
DoubleBastionAdmin authored on 09/05/2022 18:29:08
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
| ... | ... |
@@ -1,121 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
- |
|
| 8 |
-It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, but also without Asterisk, if connected directly |
|
| 9 |
-to the SIP provider (this implies that the SIP provider allows direct connections from WebRTC applications). For calls to and from regular phone numbers, a SIP provider like Telnyx or |
|
| 10 |
-Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or |
|
| 11 |
-dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory |
|
| 12 |
-and to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from |
|
| 13 |
-WebRTC applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip |
|
| 14 |
-Phone is based on the ctxSip phone. |
|
| 15 |
- |
|
| 16 |
-<span style="display:block;height:20px!important"></span> |
|
| 17 |
- |
|
| 18 |
-## Main Features |
|
| 19 |
-<span style="display:block;height:10px!important"></span> |
|
| 20 |
- |
|
| 21 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
- |
|
| 23 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
- |
|
| 25 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
- |
|
| 27 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
- |
|
| 29 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
- |
|
| 31 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
- |
|
| 33 |
-* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
- |
|
| 35 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
- |
|
| 37 |
-* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
- |
|
| 39 |
-Double Bastion is not affiliated with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
- |
|
| 41 |
-<span style="display:block;height:20px!important"></span> |
|
| 42 |
- |
|
| 43 |
-<p align="center">Initial screen</p> |
|
| 44 |
-<span style="display:block;height:10px!important"></span> |
|
| 45 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 46 |
-<span style="display:block;height:40px!important"></span> |
|
| 47 |
-<p align="center">Dialpad</p> |
|
| 48 |
-<span style="display:block;height:10px!important"></span> |
|
| 49 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
-<span style="display:block;height:40px!important"></span> |
|
| 51 |
-<p align="center">Making calls</p> |
|
| 52 |
-<span style="display:block;height:10px!important"></span> |
|
| 53 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
-<span style="display:block;height:40px!important"></span> |
|
| 55 |
-<p align="center">Transferring calls</p> |
|
| 56 |
-<span style="display:block;height:10px!important"></span> |
|
| 57 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
-<span style="display:block;height:40px!important"></span> |
|
| 59 |
- |
|
| 60 |
-## Browsers |
|
| 61 |
-<span style="display:block;height:10px!important"></span> |
|
| 62 |
- |
|
| 63 |
-SIP Trip Phone works with all the major browsers. |
|
| 64 |
- |
|
| 65 |
-<span style="display:block;height:20px!important"></span> |
|
| 66 |
- |
|
| 67 |
-## Programming Languages |
|
| 68 |
-<span style="display:block;height:10px!important"></span> |
|
| 69 |
- |
|
| 70 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 71 |
- |
|
| 72 |
-<span style="display:block;height:20px!important"></span> |
|
| 73 |
- |
|
| 74 |
-## Minimum Requirements |
|
| 75 |
-<span style="display:block;height:10px!important"></span> |
|
| 76 |
- |
|
| 77 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 78 |
- |
|
| 79 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 80 |
- |
|
| 81 |
-If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 82 |
-server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 83 |
- |
|
| 84 |
-<span style="display:block;height:20px!important"></span> |
|
| 85 |
- |
|
| 86 |
-## Installation |
|
| 87 |
-<span style="display:block;height:10px!important"></span> |
|
| 88 |
- |
|
| 89 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 90 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 91 |
- |
|
| 92 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 93 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 94 |
- |
|
| 95 |
-<span style="display:block;height:20px!important"></span> |
|
| 96 |
- |
|
| 97 |
-## Contribute |
|
| 98 |
-<span style="display:block;height:10px!important"></span> |
|
| 99 |
- |
|
| 100 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 101 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 102 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 103 |
- |
|
| 104 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 105 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 106 |
-project. |
|
| 107 |
- |
|
| 108 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 109 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 110 |
- |
|
| 111 |
-<span style="display:block;height:20px!important"></span> |
|
| 112 |
- |
|
| 113 |
-## License |
|
| 114 |
-<span style="display:block;height:10px!important"></span> |
|
| 115 |
- |
|
| 116 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 117 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 118 |
- |
|
| 119 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 120 |
- |
|
| 121 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added CHANGELOG.txt README.md appinfo/info.xml appinfo/signature.json js/launchphone.js img/sip_trip_phone_initial_screen.png phone/index.html phone/images/sip_trip_phone_logo_large.svg
DoubleBastionAdmin authored on 19/04/2022 09:04:33
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,121 @@ |
| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, but also without Asterisk, if connected directly |
|
| 9 |
+to the SIP provider (this implies that the SIP provider allows direct connections from WebRTC applications). For calls to and from regular phone numbers, a SIP provider like Telnyx or |
|
| 10 |
+Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or |
|
| 11 |
+dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory |
|
| 12 |
+and to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from |
|
| 13 |
+WebRTC applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip |
|
| 14 |
+Phone is based on the ctxSip phone. |
|
| 15 |
+ |
|
| 16 |
+<span style="display:block;height:20px!important"></span> |
|
| 17 |
+ |
|
| 18 |
+## Main Features |
|
| 19 |
+<span style="display:block;height:10px!important"></span> |
|
| 20 |
+ |
|
| 21 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
+ |
|
| 23 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
+ |
|
| 25 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
+ |
|
| 27 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
+ |
|
| 29 |
+* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
+ |
|
| 31 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
+ |
|
| 33 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
+ |
|
| 35 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
+ |
|
| 37 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
+ |
|
| 39 |
+Double Bastion is not affiliated with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
+ |
|
| 41 |
+<span style="display:block;height:20px!important"></span> |
|
| 42 |
+ |
|
| 43 |
+<p align="center">Initial screen</p> |
|
| 44 |
+<span style="display:block;height:10px!important"></span> |
|
| 45 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 46 |
+<span style="display:block;height:40px!important"></span> |
|
| 47 |
+<p align="center">Dialpad</p> |
|
| 48 |
+<span style="display:block;height:10px!important"></span> |
|
| 49 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
+<span style="display:block;height:40px!important"></span> |
|
| 51 |
+<p align="center">Making calls</p> |
|
| 52 |
+<span style="display:block;height:10px!important"></span> |
|
| 53 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
+<span style="display:block;height:40px!important"></span> |
|
| 55 |
+<p align="center">Transferring calls</p> |
|
| 56 |
+<span style="display:block;height:10px!important"></span> |
|
| 57 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
+<span style="display:block;height:40px!important"></span> |
|
| 59 |
+ |
|
| 60 |
+## Browsers |
|
| 61 |
+<span style="display:block;height:10px!important"></span> |
|
| 62 |
+ |
|
| 63 |
+SIP Trip Phone works with all the major browsers. |
|
| 64 |
+ |
|
| 65 |
+<span style="display:block;height:20px!important"></span> |
|
| 66 |
+ |
|
| 67 |
+## Programming Languages |
|
| 68 |
+<span style="display:block;height:10px!important"></span> |
|
| 69 |
+ |
|
| 70 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 71 |
+ |
|
| 72 |
+<span style="display:block;height:20px!important"></span> |
|
| 73 |
+ |
|
| 74 |
+## Minimum Requirements |
|
| 75 |
+<span style="display:block;height:10px!important"></span> |
|
| 76 |
+ |
|
| 77 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 78 |
+ |
|
| 79 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 80 |
+ |
|
| 81 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 82 |
+server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 83 |
+ |
|
| 84 |
+<span style="display:block;height:20px!important"></span> |
|
| 85 |
+ |
|
| 86 |
+## Installation |
|
| 87 |
+<span style="display:block;height:10px!important"></span> |
|
| 88 |
+ |
|
| 89 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 90 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 91 |
+ |
|
| 92 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 93 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 94 |
+ |
|
| 95 |
+<span style="display:block;height:20px!important"></span> |
|
| 96 |
+ |
|
| 97 |
+## Contribute |
|
| 98 |
+<span style="display:block;height:10px!important"></span> |
|
| 99 |
+ |
|
| 100 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 101 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 102 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 103 |
+ |
|
| 104 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 105 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 106 |
+project. |
|
| 107 |
+ |
|
| 108 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 109 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 110 |
+ |
|
| 111 |
+<span style="display:block;height:20px!important"></span> |
|
| 112 |
+ |
|
| 113 |
+## License |
|
| 114 |
+<span style="display:block;height:10px!important"></span> |
|
| 115 |
+ |
|
| 116 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 117 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 118 |
+ |
|
| 119 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 120 |
+ |
|
| 121 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed CHANGELOG.txt README.md appinfo/info.xml appinfo/signature.json js/launchphone.js img/sip_trip_phone_initial_screen.png phone/index.html
DoubleBastionAdmin authored on 19/04/2022 08:59:37
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| 1 | 1 |
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| ... | ... |
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| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
- |
|
| 8 |
-It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, but also without Asterisk, if connected directly |
|
| 9 |
-to the SIP provider (this implies that the SIP provider allows direct connections from WebRTC applications). For calls to and from regular phone numbers, a SIP provider like Telnyx or |
|
| 10 |
-Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or |
|
| 11 |
-dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory |
|
| 12 |
-and to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from |
|
| 13 |
-WebRTC applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip |
|
| 14 |
-Phone is based on the ctxSip phone. |
|
| 15 |
- |
|
| 16 |
-<span style="display:block;height:20px!important"></span> |
|
| 17 |
- |
|
| 18 |
-## Main Features |
|
| 19 |
-<span style="display:block;height:10px!important"></span> |
|
| 20 |
- |
|
| 21 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
- |
|
| 23 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
- |
|
| 25 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
- |
|
| 27 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
- |
|
| 29 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
- |
|
| 31 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
- |
|
| 33 |
-* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
- |
|
| 35 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
- |
|
| 37 |
-* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
- |
|
| 39 |
-Double Bastion is not affiliated with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
- |
|
| 41 |
-<span style="display:block;height:20px!important"></span> |
|
| 42 |
- |
|
| 43 |
-<p align="center">Initial screen</p> |
|
| 44 |
-<span style="display:block;height:10px!important"></span> |
|
| 45 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 46 |
-<span style="display:block;height:40px!important"></span> |
|
| 47 |
-<p align="center">Dialpad</p> |
|
| 48 |
-<span style="display:block;height:10px!important"></span> |
|
| 49 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
-<span style="display:block;height:40px!important"></span> |
|
| 51 |
-<p align="center">Making calls</p> |
|
| 52 |
-<span style="display:block;height:10px!important"></span> |
|
| 53 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
-<span style="display:block;height:40px!important"></span> |
|
| 55 |
-<p align="center">Transferring calls</p> |
|
| 56 |
-<span style="display:block;height:10px!important"></span> |
|
| 57 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
-<span style="display:block;height:40px!important"></span> |
|
| 59 |
- |
|
| 60 |
-## Browsers |
|
| 61 |
-<span style="display:block;height:10px!important"></span> |
|
| 62 |
- |
|
| 63 |
-SIP Trip Phone works with all the major browsers. |
|
| 64 |
- |
|
| 65 |
-<span style="display:block;height:20px!important"></span> |
|
| 66 |
- |
|
| 67 |
-## Programming Languages |
|
| 68 |
-<span style="display:block;height:10px!important"></span> |
|
| 69 |
- |
|
| 70 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 71 |
- |
|
| 72 |
-<span style="display:block;height:20px!important"></span> |
|
| 73 |
- |
|
| 74 |
-## Minimum Requirements |
|
| 75 |
-<span style="display:block;height:10px!important"></span> |
|
| 76 |
- |
|
| 77 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 78 |
- |
|
| 79 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 80 |
- |
|
| 81 |
-If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 82 |
-server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 83 |
- |
|
| 84 |
-<span style="display:block;height:20px!important"></span> |
|
| 85 |
- |
|
| 86 |
-## Installation |
|
| 87 |
-<span style="display:block;height:10px!important"></span> |
|
| 88 |
- |
|
| 89 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 90 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 91 |
- |
|
| 92 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 93 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 94 |
- |
|
| 95 |
-<span style="display:block;height:20px!important"></span> |
|
| 96 |
- |
|
| 97 |
-## Contribute |
|
| 98 |
-<span style="display:block;height:10px!important"></span> |
|
| 99 |
- |
|
| 100 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 101 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 102 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 103 |
- |
|
| 104 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 105 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 106 |
-project. |
|
| 107 |
- |
|
| 108 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 109 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 110 |
- |
|
| 111 |
-<span style="display:block;height:20px!important"></span> |
|
| 112 |
- |
|
| 113 |
-## License |
|
| 114 |
-<span style="display:block;height:10px!important"></span> |
|
| 115 |
- |
|
| 116 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 117 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 118 |
- |
|
| 119 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 120 |
- |
|
| 121 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added appinfo/info.xml appinfo/signature.json README.md
DoubleBastionAdmin authored on 16/04/2022 20:52:14
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,121 @@ |
| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a browser phone in the form of a Nextcloud application. It can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+It can be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, but also without Asterisk, if connected directly |
|
| 9 |
+to the SIP provider (this implies that the SIP provider allows direct connections from WebRTC applications). For calls to and from regular phone numbers, a SIP provider like Telnyx or |
|
| 10 |
+Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk version 18.0.0 LTS and it has to be installed on a VPS or |
|
| 11 |
+dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory |
|
| 12 |
+and to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from |
|
| 13 |
+WebRTC applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip |
|
| 14 |
+Phone is based on the ctxSip phone. |
|
| 15 |
+ |
|
| 16 |
+<span style="display:block;height:20px!important"></span> |
|
| 17 |
+ |
|
| 18 |
+## Main Features |
|
| 19 |
+<span style="display:block;height:10px!important"></span> |
|
| 20 |
+ |
|
| 21 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
+ |
|
| 23 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
+ |
|
| 25 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
+ |
|
| 27 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
+ |
|
| 29 |
+* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
+ |
|
| 31 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
+ |
|
| 33 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
+ |
|
| 35 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
+ |
|
| 37 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
+ |
|
| 39 |
+Double Bastion is not affiliated with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
+ |
|
| 41 |
+<span style="display:block;height:20px!important"></span> |
|
| 42 |
+ |
|
| 43 |
+<p align="center">Initial screen</p> |
|
| 44 |
+<span style="display:block;height:10px!important"></span> |
|
| 45 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 46 |
+<span style="display:block;height:40px!important"></span> |
|
| 47 |
+<p align="center">Dialpad</p> |
|
| 48 |
+<span style="display:block;height:10px!important"></span> |
|
| 49 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
+<span style="display:block;height:40px!important"></span> |
|
| 51 |
+<p align="center">Making calls</p> |
|
| 52 |
+<span style="display:block;height:10px!important"></span> |
|
| 53 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
+<span style="display:block;height:40px!important"></span> |
|
| 55 |
+<p align="center">Transferring calls</p> |
|
| 56 |
+<span style="display:block;height:10px!important"></span> |
|
| 57 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
+<span style="display:block;height:40px!important"></span> |
|
| 59 |
+ |
|
| 60 |
+## Browsers |
|
| 61 |
+<span style="display:block;height:10px!important"></span> |
|
| 62 |
+ |
|
| 63 |
+SIP Trip Phone works with all the major browsers. |
|
| 64 |
+ |
|
| 65 |
+<span style="display:block;height:20px!important"></span> |
|
| 66 |
+ |
|
| 67 |
+## Programming Languages |
|
| 68 |
+<span style="display:block;height:10px!important"></span> |
|
| 69 |
+ |
|
| 70 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 71 |
+ |
|
| 72 |
+<span style="display:block;height:20px!important"></span> |
|
| 73 |
+ |
|
| 74 |
+## Minimum Requirements |
|
| 75 |
+<span style="display:block;height:10px!important"></span> |
|
| 76 |
+ |
|
| 77 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 78 |
+ |
|
| 79 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 80 |
+ |
|
| 81 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 82 |
+server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 83 |
+ |
|
| 84 |
+<span style="display:block;height:20px!important"></span> |
|
| 85 |
+ |
|
| 86 |
+## Installation |
|
| 87 |
+<span style="display:block;height:10px!important"></span> |
|
| 88 |
+ |
|
| 89 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 90 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 91 |
+ |
|
| 92 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 93 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 94 |
+ |
|
| 95 |
+<span style="display:block;height:20px!important"></span> |
|
| 96 |
+ |
|
| 97 |
+## Contribute |
|
| 98 |
+<span style="display:block;height:10px!important"></span> |
|
| 99 |
+ |
|
| 100 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 101 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 102 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 103 |
+ |
|
| 104 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 105 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 106 |
+project. |
|
| 107 |
+ |
|
| 108 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 109 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 110 |
+ |
|
| 111 |
+<span style="display:block;height:20px!important"></span> |
|
| 112 |
+ |
|
| 113 |
+## License |
|
| 114 |
+<span style="display:block;height:10px!important"></span> |
|
| 115 |
+ |
|
| 116 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 117 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 118 |
+ |
|
| 119 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 120 |
+ |
|
| 121 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed appinfo/info.xml appinfo/signature.json README.md
DoubleBastionAdmin authored on 16/04/2022 20:51:12
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
| ... | ... |
@@ -1,121 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a Nextcloud application that acts like a browser phone. It uses SIP over WebSocket and WebRTC and can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
- |
|
| 8 |
-Although it's intended to be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, it can be used by itself, if it's |
|
| 9 |
-configured to connect directly to the SIP provider and the SIP provider allows direct connections from WebRTC applications. For calls to and from regular phone numbers, a SIP provider |
|
| 10 |
-like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk version 18.0.0 LTS and it has to be installed on |
|
| 11 |
-a VPS or dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific |
|
| 12 |
-directory and to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers |
|
| 13 |
-or from WebRTC applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP |
|
| 14 |
-Trip Phone is based on the ctxSip phone. |
|
| 15 |
- |
|
| 16 |
-<span style="display:block;height:20px!important"></span> |
|
| 17 |
- |
|
| 18 |
-## Main Features |
|
| 19 |
-<span style="display:block;height:10px!important"></span> |
|
| 20 |
- |
|
| 21 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
- |
|
| 23 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
- |
|
| 25 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
- |
|
| 27 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
- |
|
| 29 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
- |
|
| 31 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
- |
|
| 33 |
-* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
- |
|
| 35 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
- |
|
| 37 |
-* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
- |
|
| 39 |
-Double Bastion is not affiliated with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
- |
|
| 41 |
-<span style="display:block;height:20px!important"></span> |
|
| 42 |
- |
|
| 43 |
-<p align="center">Initial screen</p> |
|
| 44 |
-<span style="display:block;height:10px!important"></span> |
|
| 45 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 46 |
-<span style="display:block;height:40px!important"></span> |
|
| 47 |
-<p align="center">Dialpad</p> |
|
| 48 |
-<span style="display:block;height:10px!important"></span> |
|
| 49 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
-<span style="display:block;height:40px!important"></span> |
|
| 51 |
-<p align="center">Making calls</p> |
|
| 52 |
-<span style="display:block;height:10px!important"></span> |
|
| 53 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
-<span style="display:block;height:40px!important"></span> |
|
| 55 |
-<p align="center">Transferring calls</p> |
|
| 56 |
-<span style="display:block;height:10px!important"></span> |
|
| 57 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
-<span style="display:block;height:40px!important"></span> |
|
| 59 |
- |
|
| 60 |
-## Browsers |
|
| 61 |
-<span style="display:block;height:10px!important"></span> |
|
| 62 |
- |
|
| 63 |
-SIP Trip Phone works with all the major browsers. |
|
| 64 |
- |
|
| 65 |
-<span style="display:block;height:20px!important"></span> |
|
| 66 |
- |
|
| 67 |
-## Programming Languages |
|
| 68 |
-<span style="display:block;height:10px!important"></span> |
|
| 69 |
- |
|
| 70 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 71 |
- |
|
| 72 |
-<span style="display:block;height:20px!important"></span> |
|
| 73 |
- |
|
| 74 |
-## Minimum Requirements |
|
| 75 |
-<span style="display:block;height:10px!important"></span> |
|
| 76 |
- |
|
| 77 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 78 |
- |
|
| 79 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 80 |
- |
|
| 81 |
-If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 82 |
-server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 83 |
- |
|
| 84 |
-<span style="display:block;height:20px!important"></span> |
|
| 85 |
- |
|
| 86 |
-## Installation |
|
| 87 |
-<span style="display:block;height:10px!important"></span> |
|
| 88 |
- |
|
| 89 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 90 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 91 |
- |
|
| 92 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 93 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 94 |
- |
|
| 95 |
-<span style="display:block;height:20px!important"></span> |
|
| 96 |
- |
|
| 97 |
-## Contribute |
|
| 98 |
-<span style="display:block;height:10px!important"></span> |
|
| 99 |
- |
|
| 100 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 101 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 102 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 103 |
- |
|
| 104 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 105 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 106 |
-project. |
|
| 107 |
- |
|
| 108 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 109 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 110 |
- |
|
| 111 |
-<span style="display:block;height:20px!important"></span> |
|
| 112 |
- |
|
| 113 |
-## License |
|
| 114 |
-<span style="display:block;height:10px!important"></span> |
|
| 115 |
- |
|
| 116 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 117 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 118 |
- |
|
| 119 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 120 |
- |
|
| 121 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added appinfo/info.xml appinfo/signature.json README.md
DoubleBastionAdmin authored on 16/04/2022 20:11:54
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,121 @@ |
| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a Nextcloud application that acts like a browser phone. It uses SIP over WebSocket and WebRTC and can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+Although it's intended to be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, it can be used by itself, if it's |
|
| 9 |
+configured to connect directly to the SIP provider and the SIP provider allows direct connections from WebRTC applications. For calls to and from regular phone numbers, a SIP provider |
|
| 10 |
+like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk version 18.0.0 LTS and it has to be installed on |
|
| 11 |
+a VPS or dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific |
|
| 12 |
+directory and to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers |
|
| 13 |
+or from WebRTC applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP |
|
| 14 |
+Trip Phone is based on the ctxSip phone. |
|
| 15 |
+ |
|
| 16 |
+<span style="display:block;height:20px!important"></span> |
|
| 17 |
+ |
|
| 18 |
+## Main Features |
|
| 19 |
+<span style="display:block;height:10px!important"></span> |
|
| 20 |
+ |
|
| 21 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
+ |
|
| 23 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
+ |
|
| 25 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
+ |
|
| 27 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
+ |
|
| 29 |
+* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
+ |
|
| 31 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
+ |
|
| 33 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
+ |
|
| 35 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
+ |
|
| 37 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
+ |
|
| 39 |
+Double Bastion is not affiliated with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
+ |
|
| 41 |
+<span style="display:block;height:20px!important"></span> |
|
| 42 |
+ |
|
| 43 |
+<p align="center">Initial screen</p> |
|
| 44 |
+<span style="display:block;height:10px!important"></span> |
|
| 45 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 46 |
+<span style="display:block;height:40px!important"></span> |
|
| 47 |
+<p align="center">Dialpad</p> |
|
| 48 |
+<span style="display:block;height:10px!important"></span> |
|
| 49 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
+<span style="display:block;height:40px!important"></span> |
|
| 51 |
+<p align="center">Making calls</p> |
|
| 52 |
+<span style="display:block;height:10px!important"></span> |
|
| 53 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
+<span style="display:block;height:40px!important"></span> |
|
| 55 |
+<p align="center">Transferring calls</p> |
|
| 56 |
+<span style="display:block;height:10px!important"></span> |
|
| 57 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
+<span style="display:block;height:40px!important"></span> |
|
| 59 |
+ |
|
| 60 |
+## Browsers |
|
| 61 |
+<span style="display:block;height:10px!important"></span> |
|
| 62 |
+ |
|
| 63 |
+SIP Trip Phone works with all the major browsers. |
|
| 64 |
+ |
|
| 65 |
+<span style="display:block;height:20px!important"></span> |
|
| 66 |
+ |
|
| 67 |
+## Programming Languages |
|
| 68 |
+<span style="display:block;height:10px!important"></span> |
|
| 69 |
+ |
|
| 70 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 71 |
+ |
|
| 72 |
+<span style="display:block;height:20px!important"></span> |
|
| 73 |
+ |
|
| 74 |
+## Minimum Requirements |
|
| 75 |
+<span style="display:block;height:10px!important"></span> |
|
| 76 |
+ |
|
| 77 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 78 |
+ |
|
| 79 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 80 |
+ |
|
| 81 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version 18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 82 |
+server. You can also install Coturn (version 4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 83 |
+ |
|
| 84 |
+<span style="display:block;height:20px!important"></span> |
|
| 85 |
+ |
|
| 86 |
+## Installation |
|
| 87 |
+<span style="display:block;height:10px!important"></span> |
|
| 88 |
+ |
|
| 89 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 90 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 91 |
+ |
|
| 92 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 93 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 94 |
+ |
|
| 95 |
+<span style="display:block;height:20px!important"></span> |
|
| 96 |
+ |
|
| 97 |
+## Contribute |
|
| 98 |
+<span style="display:block;height:10px!important"></span> |
|
| 99 |
+ |
|
| 100 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 101 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 102 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 103 |
+ |
|
| 104 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 105 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 106 |
+project. |
|
| 107 |
+ |
|
| 108 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 109 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 110 |
+ |
|
| 111 |
+<span style="display:block;height:20px!important"></span> |
|
| 112 |
+ |
|
| 113 |
+## License |
|
| 114 |
+<span style="display:block;height:10px!important"></span> |
|
| 115 |
+ |
|
| 116 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 117 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 118 |
+ |
|
| 119 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 120 |
+ |
|
| 121 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed appinfo/info.xml appinfo/signature.json README.md
DoubleBastionAdmin authored on 16/04/2022 20:09:52
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
| ... | ... |
@@ -1,125 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a Nextcloud application that acts like a browser phone. It uses SIP over WebSocket and WebRTC and can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
- |
|
| 8 |
-Although it's intended to be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, it can be used by itself, if it's |
|
| 9 |
-configured to connect directly to the SIP provider and the SIP provider allows direct connections from WebRTC applications. For calls to and from regular phone numbers, a SIP provider |
|
| 10 |
-like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk v18.0.0 LTS and it has to be installed on a VPS |
|
| 11 |
-or dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory and |
|
| 12 |
-to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from WebRTC |
|
| 13 |
-applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip Phone is |
|
| 14 |
-based on the ctxSip phone. |
|
| 15 |
- |
|
| 16 |
-<span style="display:block;height:20px!important"></span> |
|
| 17 |
- |
|
| 18 |
-## Main Features |
|
| 19 |
-<span style="display:block;height:10px!important"></span> |
|
| 20 |
- |
|
| 21 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
- |
|
| 23 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
- |
|
| 25 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
- |
|
| 27 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
- |
|
| 29 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
- |
|
| 31 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
- |
|
| 33 |
-* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
- |
|
| 35 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
- |
|
| 37 |
-* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
- |
|
| 39 |
-We have no affiliation with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
- |
|
| 41 |
-<span style="display:block;height:10px!important"></span> |
|
| 42 |
- |
|
| 43 |
-Double Bastion is not affiliated with Telnyx or Localphone. |
|
| 44 |
- |
|
| 45 |
-<span style="display:block;height:20px!important"></span> |
|
| 46 |
- |
|
| 47 |
-<p align="center">Initial screen</p> |
|
| 48 |
-<span style="display:block;height:10px!important"></span> |
|
| 49 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
-<span style="display:block;height:40px!important"></span> |
|
| 51 |
-<p align="center">Dialpad</p> |
|
| 52 |
-<span style="display:block;height:10px!important"></span> |
|
| 53 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
-<span style="display:block;height:40px!important"></span> |
|
| 55 |
-<p align="center">Making calls</p> |
|
| 56 |
-<span style="display:block;height:10px!important"></span> |
|
| 57 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
-<span style="display:block;height:40px!important"></span> |
|
| 59 |
-<p align="center">Transferring calls</p> |
|
| 60 |
-<span style="display:block;height:10px!important"></span> |
|
| 61 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 62 |
-<span style="display:block;height:40px!important"></span> |
|
| 63 |
- |
|
| 64 |
-## Browsers |
|
| 65 |
-<span style="display:block;height:10px!important"></span> |
|
| 66 |
- |
|
| 67 |
-SIP Trip Phone works with all the major browsers. |
|
| 68 |
- |
|
| 69 |
-<span style="display:block;height:20px!important"></span> |
|
| 70 |
- |
|
| 71 |
-## Programming Languages |
|
| 72 |
-<span style="display:block;height:10px!important"></span> |
|
| 73 |
- |
|
| 74 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 75 |
- |
|
| 76 |
-<span style="display:block;height:20px!important"></span> |
|
| 77 |
- |
|
| 78 |
-## Minimum Requirements |
|
| 79 |
-<span style="display:block;height:10px!important"></span> |
|
| 80 |
- |
|
| 81 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 82 |
- |
|
| 83 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 84 |
- |
|
| 85 |
-If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version v18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 86 |
-server. You can also install Coturn (v4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 87 |
- |
|
| 88 |
-<span style="display:block;height:20px!important"></span> |
|
| 89 |
- |
|
| 90 |
-## Installation |
|
| 91 |
-<span style="display:block;height:10px!important"></span> |
|
| 92 |
- |
|
| 93 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 94 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 95 |
- |
|
| 96 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 97 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 98 |
- |
|
| 99 |
-<span style="display:block;height:20px!important"></span> |
|
| 100 |
- |
|
| 101 |
-## Contribute |
|
| 102 |
-<span style="display:block;height:10px!important"></span> |
|
| 103 |
- |
|
| 104 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 105 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 106 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 107 |
- |
|
| 108 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 109 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 110 |
-project. |
|
| 111 |
- |
|
| 112 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 113 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 114 |
- |
|
| 115 |
-<span style="display:block;height:20px!important"></span> |
|
| 116 |
- |
|
| 117 |
-## License |
|
| 118 |
-<span style="display:block;height:10px!important"></span> |
|
| 119 |
- |
|
| 120 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 121 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 122 |
- |
|
| 123 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 124 |
- |
|
| 125 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added appinfo/info.xml appinfo/signature.json README.md
DoubleBastionAdmin authored on 16/04/2022 19:45:50
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
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| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a Nextcloud application that acts like a browser phone. It uses SIP over WebSocket and WebRTC and can connect to SIP providers via Asterisk or directly.** |
|
| 7 |
+ |
|
| 8 |
+Although it's intended to be used in conjunction with Asterisk, to benefit from the control, autonomy and advanced PBX features offered by Asterisk, it can be used by itself, if it's |
|
| 9 |
+configured to connect directly to the SIP provider and the SIP provider allows direct connections from WebRTC applications. For calls to and from regular phone numbers, a SIP provider |
|
| 10 |
+like Telnyx or Localphone is needed and a real phone number acquired from that SIP provider. If Asterisk is used, it has to be Asterisk v18.0.0 LTS and it has to be installed on a VPS |
|
| 11 |
+or dedicated server, as explained in the documentation mentioned in the 'Installation' section from below. The web server has to be configured to allow access to a specific directory and |
|
| 12 |
+to proxy WebSocket traffic to a specific URL, as explained in detail in the documentation. Not all SIP providers allow direct connections from external Asterisk servers or from WebRTC |
|
| 13 |
+applications. Telnyx and Localphone allow direct connections from their clients' Asterisk servers and Telnyx also allows direct connections from WebRTC applications. SIP Trip Phone is |
|
| 14 |
+based on the ctxSip phone. |
|
| 15 |
+ |
|
| 16 |
+<span style="display:block;height:20px!important"></span> |
|
| 17 |
+ |
|
| 18 |
+## Main Features |
|
| 19 |
+<span style="display:block;height:10px!important"></span> |
|
| 20 |
+ |
|
| 21 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can cost even 90% less than regular phone calls. |
|
| 22 |
+ |
|
| 23 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 24 |
+ |
|
| 25 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 26 |
+ |
|
| 27 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 28 |
+ |
|
| 29 |
+* 🚩 On-screen notifications on incoming calls. |
|
| 30 |
+ |
|
| 31 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 32 |
+ |
|
| 33 |
+* 💻 If Asterisk is used, on the underlying Asterisk server you can implement an IVR (Interactive Voice Response or 'voice menu') and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 34 |
+ |
|
| 35 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 36 |
+ |
|
| 37 |
+* 💸 Low per minute prices: if Asterisk is used, you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). If SIP Trip Phone is connected directly to Telnyx, you can make and receive phone calls with $0.0020 per minute in the US. |
|
| 38 |
+ |
|
| 39 |
+We have no affiliation with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them as SIP providers. |
|
| 40 |
+ |
|
| 41 |
+<span style="display:block;height:10px!important"></span> |
|
| 42 |
+ |
|
| 43 |
+Double Bastion is not affiliated with Telnyx or Localphone. |
|
| 44 |
+ |
|
| 45 |
+<span style="display:block;height:20px!important"></span> |
|
| 46 |
+ |
|
| 47 |
+<p align="center">Initial screen</p> |
|
| 48 |
+<span style="display:block;height:10px!important"></span> |
|
| 49 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 50 |
+<span style="display:block;height:40px!important"></span> |
|
| 51 |
+<p align="center">Dialpad</p> |
|
| 52 |
+<span style="display:block;height:10px!important"></span> |
|
| 53 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 54 |
+<span style="display:block;height:40px!important"></span> |
|
| 55 |
+<p align="center">Making calls</p> |
|
| 56 |
+<span style="display:block;height:10px!important"></span> |
|
| 57 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 58 |
+<span style="display:block;height:40px!important"></span> |
|
| 59 |
+<p align="center">Transferring calls</p> |
|
| 60 |
+<span style="display:block;height:10px!important"></span> |
|
| 61 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 62 |
+<span style="display:block;height:40px!important"></span> |
|
| 63 |
+ |
|
| 64 |
+## Browsers |
|
| 65 |
+<span style="display:block;height:10px!important"></span> |
|
| 66 |
+ |
|
| 67 |
+SIP Trip Phone works with all the major browsers. |
|
| 68 |
+ |
|
| 69 |
+<span style="display:block;height:20px!important"></span> |
|
| 70 |
+ |
|
| 71 |
+## Programming Languages |
|
| 72 |
+<span style="display:block;height:10px!important"></span> |
|
| 73 |
+ |
|
| 74 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 75 |
+ |
|
| 76 |
+<span style="display:block;height:20px!important"></span> |
|
| 77 |
+ |
|
| 78 |
+## Minimum Requirements |
|
| 79 |
+<span style="display:block;height:10px!important"></span> |
|
| 80 |
+ |
|
| 81 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 82 |
+ |
|
| 83 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. You can use a SIP provider different from Telnyx or Localphone, but they have to allow direct connections from external Asterisk servers and/or from WebRTC applications. |
|
| 84 |
+ |
|
| 85 |
+If you decide to connect SIP Trip Phone to your SIP provider via Asterisk, you will need **Asterisk version v18.0.0 LTS** (with **chan_pjsip** enabled), installed on a VPS or dedicated |
|
| 86 |
+server. You can also install Coturn (v4.5.1.1 or newer) as a STUN server, which facilitates connections when callers are behind routers. |
|
| 87 |
+ |
|
| 88 |
+<span style="display:block;height:20px!important"></span> |
|
| 89 |
+ |
|
| 90 |
+## Installation |
|
| 91 |
+<span style="display:block;height:10px!important"></span> |
|
| 92 |
+ |
|
| 93 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 94 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 95 |
+ |
|
| 96 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 97 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 98 |
+ |
|
| 99 |
+<span style="display:block;height:20px!important"></span> |
|
| 100 |
+ |
|
| 101 |
+## Contribute |
|
| 102 |
+<span style="display:block;height:10px!important"></span> |
|
| 103 |
+ |
|
| 104 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 105 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 106 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 107 |
+ |
|
| 108 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 109 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes, we can include them in the |
|
| 110 |
+project. |
|
| 111 |
+ |
|
| 112 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 113 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 114 |
+ |
|
| 115 |
+<span style="display:block;height:20px!important"></span> |
|
| 116 |
+ |
|
| 117 |
+## License |
|
| 118 |
+<span style="display:block;height:10px!important"></span> |
|
| 119 |
+ |
|
| 120 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 121 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 122 |
+ |
|
| 123 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 124 |
+ |
|
| 125 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed appinfo/info.xml appinfo/signature.json README.md
DoubleBastionAdmin authored on 16/04/2022 19:44:32
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
| ... | ... |
@@ -1,123 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a Nextcloud application that acts like a browser phone. It connects to an Asterisk server to make and receive phone calls using SIP over WebSocket and WebRTC.** |
|
| 7 |
- |
|
| 8 |
-For calls to and from regular phone numbers, a SIP provider like Telnyx or Localphone is needed and a real phone number acquired from the SIP provider. Asterisk must be installed |
|
| 9 |
-on a VPS or dedicated server. It is assumed that Nextcloud uses HTTPS and that it is served on a subdomain, like 'cloud.example.com', and not on a subdirectory, like |
|
| 10 |
-'example.com/nextcloud/'. Once SIP Trip Phone gets connected to Asterisk, Asterisk can be connected to any SIP provider, but SIP Trip Phone has been tested only with Telnyx and |
|
| 11 |
-Localphone. This application is based on the ctxSip phone. |
|
| 12 |
- |
|
| 13 |
-<span style="display:block;height:20px!important"></span> |
|
| 14 |
- |
|
| 15 |
-## Main Features |
|
| 16 |
-<span style="display:block;height:10px!important"></span> |
|
| 17 |
- |
|
| 18 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can be even 90% cheaper. |
|
| 19 |
- |
|
| 20 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 21 |
- |
|
| 22 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 23 |
- |
|
| 24 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 25 |
- |
|
| 26 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 27 |
- |
|
| 28 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 29 |
- |
|
| 30 |
-* 💻 On the underlying Asterisk server you can implement an IVR (Interactive Voice Response or voice menu) and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 31 |
- |
|
| 32 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 33 |
- |
|
| 34 |
-* 💸 Low per minute prices: you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). |
|
| 35 |
- |
|
| 36 |
-We have no affiliation with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them. |
|
| 37 |
- |
|
| 38 |
-<span style="display:block;height:10px!important"></span> |
|
| 39 |
- |
|
| 40 |
-Double Bastion is not affiliated with Telnyx or Localphone. |
|
| 41 |
- |
|
| 42 |
-<span style="display:block;height:20px!important"></span> |
|
| 43 |
- |
|
| 44 |
-<p align="center">Initial screen</p> |
|
| 45 |
-<span style="display:block;height:10px!important"></span> |
|
| 46 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 47 |
-<span style="display:block;height:40px!important"></span> |
|
| 48 |
-<p align="center">Dialpad</p> |
|
| 49 |
-<span style="display:block;height:10px!important"></span> |
|
| 50 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 51 |
-<span style="display:block;height:40px!important"></span> |
|
| 52 |
-<p align="center">Making calls</p> |
|
| 53 |
-<span style="display:block;height:10px!important"></span> |
|
| 54 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 55 |
-<span style="display:block;height:40px!important"></span> |
|
| 56 |
-<p align="center">Transferring calls</p> |
|
| 57 |
-<span style="display:block;height:10px!important"></span> |
|
| 58 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 59 |
-<span style="display:block;height:40px!important"></span> |
|
| 60 |
- |
|
| 61 |
-## Browsers |
|
| 62 |
-<span style="display:block;height:10px!important"></span> |
|
| 63 |
- |
|
| 64 |
-SIP Trip Phone works with all the major browsers. |
|
| 65 |
- |
|
| 66 |
-<span style="display:block;height:20px!important"></span> |
|
| 67 |
- |
|
| 68 |
-## Programming Languages |
|
| 69 |
-<span style="display:block;height:10px!important"></span> |
|
| 70 |
- |
|
| 71 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 72 |
- |
|
| 73 |
-<span style="display:block;height:20px!important"></span> |
|
| 74 |
- |
|
| 75 |
-## Minimum Requirements |
|
| 76 |
-<span style="display:block;height:10px!important"></span> |
|
| 77 |
- |
|
| 78 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 79 |
- |
|
| 80 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. |
|
| 81 |
- |
|
| 82 |
-- **Asterisk** (with **chan_pjsip** enabled) installed on a VPS or dedicated server. |
|
| 83 |
- |
|
| 84 |
-You can also install **Coturn** as a STUN server, which helps when callers are behind routers. |
|
| 85 |
- |
|
| 86 |
-<span style="display:block;height:20px!important"></span> |
|
| 87 |
- |
|
| 88 |
-## Installation |
|
| 89 |
-<span style="display:block;height:10px!important"></span> |
|
| 90 |
- |
|
| 91 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 92 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 93 |
- |
|
| 94 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 95 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 96 |
- |
|
| 97 |
-<span style="display:block;height:20px!important"></span> |
|
| 98 |
- |
|
| 99 |
-## Contribute |
|
| 100 |
-<span style="display:block;height:10px!important"></span> |
|
| 101 |
- |
|
| 102 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 103 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 104 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 105 |
- |
|
| 106 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 107 |
-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes we can include them in the |
|
| 108 |
-project. |
|
| 109 |
- |
|
| 110 |
-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 111 |
-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 112 |
- |
|
| 113 |
-<span style="display:block;height:20px!important"></span> |
|
| 114 |
- |
|
| 115 |
-## License |
|
| 116 |
-<span style="display:block;height:10px!important"></span> |
|
| 117 |
- |
|
| 118 |
-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 119 |
-the terms of the GNU Affero General Public License Version 3. |
|
| 120 |
- |
|
| 121 |
-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 122 |
- |
|
| 123 |
-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
added appinfo/info.xml appinfo/signature.json CHANGELOG.txt README.md
DoubleBastionAdmin authored on 30/03/2022 20:42:28
Showing 1 changed files
| 1 | 1 |
new file mode 100644 |
| ... | ... |
@@ -0,0 +1,123 @@ |
| 1 |
+<span style="display:block;height:15px!important"></span> |
|
| 2 |
+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
+ |
|
| 4 |
+<span style="display:block;height:20px!important"></span> |
|
| 5 |
+ |
|
| 6 |
+**SIP Trip Phone is a Nextcloud application that acts like a browser phone. It connects to an Asterisk server to make and receive phone calls using SIP over WebSocket and WebRTC.** |
|
| 7 |
+ |
|
| 8 |
+For calls to and from regular phone numbers, a SIP provider like Telnyx or Localphone is needed and a real phone number acquired from the SIP provider. Asterisk must be installed |
|
| 9 |
+on a VPS or dedicated server. It is assumed that Nextcloud uses HTTPS and that it is served on a subdomain, like 'cloud.example.com', and not on a subdirectory, like |
|
| 10 |
+'example.com/nextcloud/'. Once SIP Trip Phone gets connected to Asterisk, Asterisk can be connected to any SIP provider, but SIP Trip Phone has been tested only with Telnyx and |
|
| 11 |
+Localphone. This application is based on the ctxSip phone. |
|
| 12 |
+ |
|
| 13 |
+<span style="display:block;height:20px!important"></span> |
|
| 14 |
+ |
|
| 15 |
+## Main Features |
|
| 16 |
+<span style="display:block;height:10px!important"></span> |
|
| 17 |
+ |
|
| 18 |
+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can be even 90% cheaper. |
|
| 19 |
+ |
|
| 20 |
+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 21 |
+ |
|
| 22 |
+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 23 |
+ |
|
| 24 |
+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 25 |
+ |
|
| 26 |
+* 🚩 On-screen notifications on incoming calls. |
|
| 27 |
+ |
|
| 28 |
+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 29 |
+ |
|
| 30 |
+* 💻 On the underlying Asterisk server you can implement an IVR (Interactive Voice Response or voice menu) and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 31 |
+ |
|
| 32 |
+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 33 |
+ |
|
| 34 |
+* 💸 Low per minute prices: you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). |
|
| 35 |
+ |
|
| 36 |
+We have no affiliation with Telnyx or Localphone. The quality of services and low prices are the only reasons for choosing them. |
|
| 37 |
+ |
|
| 38 |
+<span style="display:block;height:10px!important"></span> |
|
| 39 |
+ |
|
| 40 |
+Double Bastion is not affiliated with Telnyx or Localphone. |
|
| 41 |
+ |
|
| 42 |
+<span style="display:block;height:20px!important"></span> |
|
| 43 |
+ |
|
| 44 |
+<p align="center">Initial screen</p> |
|
| 45 |
+<span style="display:block;height:10px!important"></span> |
|
| 46 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 47 |
+<span style="display:block;height:40px!important"></span> |
|
| 48 |
+<p align="center">Dialpad</p> |
|
| 49 |
+<span style="display:block;height:10px!important"></span> |
|
| 50 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 51 |
+<span style="display:block;height:40px!important"></span> |
|
| 52 |
+<p align="center">Making calls</p> |
|
| 53 |
+<span style="display:block;height:10px!important"></span> |
|
| 54 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 55 |
+<span style="display:block;height:40px!important"></span> |
|
| 56 |
+<p align="center">Transferring calls</p> |
|
| 57 |
+<span style="display:block;height:10px!important"></span> |
|
| 58 |
+<span style="display:block;margin:auto;width:412px;"></span> |
|
| 59 |
+<span style="display:block;height:40px!important"></span> |
|
| 60 |
+ |
|
| 61 |
+## Browsers |
|
| 62 |
+<span style="display:block;height:10px!important"></span> |
|
| 63 |
+ |
|
| 64 |
+SIP Trip Phone works with all the major browsers. |
|
| 65 |
+ |
|
| 66 |
+<span style="display:block;height:20px!important"></span> |
|
| 67 |
+ |
|
| 68 |
+## Programming Languages |
|
| 69 |
+<span style="display:block;height:10px!important"></span> |
|
| 70 |
+ |
|
| 71 |
+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 72 |
+ |
|
| 73 |
+<span style="display:block;height:20px!important"></span> |
|
| 74 |
+ |
|
| 75 |
+## Minimum Requirements |
|
| 76 |
+<span style="display:block;height:10px!important"></span> |
|
| 77 |
+ |
|
| 78 |
+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 79 |
+ |
|
| 80 |
+- **A telnyx.com or localphone.com account and a phone number** associated with it. |
|
| 81 |
+ |
|
| 82 |
+- **Asterisk** (with **chan_pjsip** enabled) installed on a VPS or dedicated server. |
|
| 83 |
+ |
|
| 84 |
+You can also install **Coturn** as a STUN server, which helps when callers are behind routers. |
|
| 85 |
+ |
|
| 86 |
+<span style="display:block;height:20px!important"></span> |
|
| 87 |
+ |
|
| 88 |
+## Installation |
|
| 89 |
+<span style="display:block;height:10px!important"></span> |
|
| 90 |
+ |
|
| 91 |
+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 92 |
+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 93 |
+ |
|
| 94 |
+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 95 |
+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 96 |
+ |
|
| 97 |
+<span style="display:block;height:20px!important"></span> |
|
| 98 |
+ |
|
| 99 |
+## Contribute |
|
| 100 |
+<span style="display:block;height:10px!important"></span> |
|
| 101 |
+ |
|
| 102 |
+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 103 |
+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 104 |
+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 105 |
+ |
|
| 106 |
+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
| 107 |
+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes we can include them in the |
|
| 108 |
+project. |
|
| 109 |
+ |
|
| 110 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 111 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 112 |
+ |
|
| 113 |
+<span style="display:block;height:20px!important"></span> |
|
| 114 |
+ |
|
| 115 |
+## License |
|
| 116 |
+<span style="display:block;height:10px!important"></span> |
|
| 117 |
+ |
|
| 118 |
+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
|
| 119 |
+the terms of the GNU Affero General Public License Version 3. |
|
| 120 |
+ |
|
| 121 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 122 |
+ |
|
| 123 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
removed appinfo/info.xml appinfo/signature.json CHANGELOG.txt README.md
DoubleBastionAdmin authored on 30/03/2022 20:39:41
Showing 1 changed files
| 1 | 1 |
deleted file mode 100644 |
| ... | ... |
@@ -1,120 +0,0 @@ |
| 1 |
-<span style="display:block;height:15px!important"></span> |
|
| 2 |
-<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
|
| 3 |
- |
|
| 4 |
-<span style="display:block;height:20px!important"></span> |
|
| 5 |
- |
|
| 6 |
-**SIP Trip Phone is a Nextcloud application that acts like a browser phone. It connects to an Asterisk server to make and receive phone calls using SIP over WebSocket and WebRTC.** |
|
| 7 |
- |
|
| 8 |
-For calls to and from regular phone numbers, a telnyx.com or localphone.com account is needed and a real phone number acquired from one of the two providers of SIP services. |
|
| 9 |
-Nextcloud must use HTTPS. Asterisk has to be installed on a VPS or dedicated server. Once SIP Trip Phone gets connected to Asterisk, Asterisk can be connected to any SIP provider, |
|
| 10 |
-but SIP Trip Phone has been tested only with Telnyx and Localphone. This application is based on the ctxSip phone. |
|
| 11 |
- |
|
| 12 |
-<span style="display:block;height:20px!important"></span> |
|
| 13 |
- |
|
| 14 |
-## Main Features |
|
| 15 |
-<span style="display:block;height:10px!important"></span> |
|
| 16 |
- |
|
| 17 |
-* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can be even 90% cheaper. |
|
| 18 |
- |
|
| 19 |
-* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
|
| 20 |
- |
|
| 21 |
-* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
|
| 22 |
- |
|
| 23 |
-* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
|
| 24 |
- |
|
| 25 |
-* 🚩 On-screen notifications on incoming calls. |
|
| 26 |
- |
|
| 27 |
-* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
|
| 28 |
- |
|
| 29 |
-* 💻 On the underlying Asterisk server you can implement an IVR (Interactive Voice Response or voice menu) and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
|
| 30 |
- |
|
| 31 |
-* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
|
| 32 |
- |
|
| 33 |
-* 💸 Low per minute prices: you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). |
|
| 34 |
- |
|
| 35 |
-<span style="display:block;height:10px!important"></span> |
|
| 36 |
- |
|
| 37 |
-Double Bastion is not affiliated with Telnyx or Localphone. |
|
| 38 |
- |
|
| 39 |
-<span style="display:block;height:20px!important"></span> |
|
| 40 |
- |
|
| 41 |
-<p align="center">Initial screen</p> |
|
| 42 |
-<span style="display:block;height:10px!important"></span> |
|
| 43 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 44 |
-<span style="display:block;height:40px!important"></span> |
|
| 45 |
-<p align="center">Dialpad</p> |
|
| 46 |
-<span style="display:block;height:10px!important"></span> |
|
| 47 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 48 |
-<span style="display:block;height:40px!important"></span> |
|
| 49 |
-<p align="center">Making calls</p> |
|
| 50 |
-<span style="display:block;height:10px!important"></span> |
|
| 51 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 52 |
-<span style="display:block;height:40px!important"></span> |
|
| 53 |
-<p align="center">Transferring calls</p> |
|
| 54 |
-<span style="display:block;height:10px!important"></span> |
|
| 55 |
-<span style="display:block;margin:auto;width:412px;"></span> |
|
| 56 |
-<span style="display:block;height:40px!important"></span> |
|
| 57 |
- |
|
| 58 |
-## Browsers |
|
| 59 |
-<span style="display:block;height:10px!important"></span> |
|
| 60 |
- |
|
| 61 |
-SIP Trip Phone works with all the major browsers. |
|
| 62 |
- |
|
| 63 |
-<span style="display:block;height:20px!important"></span> |
|
| 64 |
- |
|
| 65 |
-## Programming Languages |
|
| 66 |
-<span style="display:block;height:10px!important"></span> |
|
| 67 |
- |
|
| 68 |
-SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
|
| 69 |
- |
|
| 70 |
-<span style="display:block;height:20px!important"></span> |
|
| 71 |
- |
|
| 72 |
-## Minimum Requirements |
|
| 73 |
-<span style="display:block;height:10px!important"></span> |
|
| 74 |
- |
|
| 75 |
-- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
|
| 76 |
- |
|
| 77 |
-- **A telnyx.com or localphone.com account and a phone number** associated with it. |
|
| 78 |
- |
|
| 79 |
-- **Asterisk** (with **chan_pjsip** enabled) installed on a VPS or dedicated server. |
|
| 80 |
- |
|
| 81 |
-You can also install **Coturn** as a STUN server, which helps when callers are behind routers. |
|
| 82 |
- |
|
| 83 |
-<span style="display:block;height:20px!important"></span> |
|
| 84 |
- |
|
| 85 |
-## Installation |
|
| 86 |
-<span style="display:block;height:10px!important"></span> |
|
| 87 |
- |
|
| 88 |
-<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
|
| 89 |
-explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
|
| 90 |
- |
|
| 91 |
-SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
|
| 92 |
-the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
|
| 93 |
- |
|
| 94 |
-<span style="display:block;height:20px!important"></span> |
|
| 95 |
- |
|
| 96 |
-## Contribute |
|
| 97 |
-<span style="display:block;height:10px!important"></span> |
|
| 98 |
- |
|
| 99 |
-This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
|
| 100 |
-repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
|
| 101 |
-digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
|
| 102 |
- |
|
| 103 |
-If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
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-mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes we can include them in the |
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-project. |
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- |
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-Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
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-issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
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- |
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-<span style="display:block;height:20px!important"></span> |
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- |
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-## License |
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-<span style="display:block;height:10px!important"></span> |
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- |
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-SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
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-the terms of the GNU Affero General Public License Version 3. |
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- |
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-This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
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- |
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-SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |
Browse code
Created repository.
DoubleBastionAdmin authored on 02/03/2022 00:26:46
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+<span style="display:block;height:15px!important"></span> |
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+<p align="center"><img src="https://git.doublebastion.com/sip-trip-phone/raw/develop/img/sip_trip_phone_logo.png" alt="SIP Trip Phone" width="171px" height="119px"/></p> |
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+ |
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+<span style="display:block;height:20px!important"></span> |
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+ |
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+**SIP Trip Phone is a Nextcloud application that acts like a browser phone. It connects to an Asterisk server to make and receive phone calls using SIP over WebSocket and WebRTC.** |
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+ |
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+For calls to and from regular phone numbers, a telnyx.com or localphone.com account is needed and a real phone number acquired from one of the two providers of SIP services. |
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+Nextcloud must use HTTPS. Asterisk has to be installed on a VPS or dedicated server. Once SIP Trip Phone gets connected to Asterisk, Asterisk can be connected to any SIP provider, |
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+but SIP Trip Phone has been tested only with Telnyx and Localphone. This application is based on the ctxSip phone. |
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+ |
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+<span style="display:block;height:20px!important"></span> |
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+ |
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+## Main Features |
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+<span style="display:block;height:10px!important"></span> |
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+ |
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+* 📞 SIP Trip Phone allows making and receiving calls to/from any mobile or landline phone at lower rates than with regular phones. It is known that VoIP phone calls are up to 70% cheaper than regular phone calls. International VoIP phone calls can be even 90% cheaper. |
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+ |
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+* 🌐 You can acquire phone numbers in countries of your choice and make cheap international phone calls to receivers in those countries. When calling you back on those numbers, the receivers will pay as for local calls. |
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+ |
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+* 🆓 You can make free calls over the Internet between extensions configured on the underlying Asterisk server. |
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+ |
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+* ☎️ SIP Trip Phone logs recent phone calls and their duration and allows pausing, muting and transferring phone calls. |
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+ |
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+* 🚩 On-screen notifications on incoming calls. |
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+ |
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+* 📃 Once you open SIP Trip Phone, you can use it even if you are logged out of Nextcloud. |
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+ |
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+* 💻 On the underlying Asterisk server you can implement an IVR (Interactive Voice Response or voice menu) and many advanced PBX features such as voicemail, queue management, music on hold, number blacklisting, call recording, audio conference calls, etc. |
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+ |
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+* 💰 The only ongoing cost is about $1 per month (depending on the country) for a phone number. No contracts. |
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+ |
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+* 💸 Low per minute prices: you can make calls within the US starting from $0.0050 per minute and receive calls with $0.0075 per minute or less (Telnyx), or $0.0060 per minute for outgoing calls and $0 for incoming calls (Localphone). |
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+ |
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+<span style="display:block;height:10px!important"></span> |
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+ |
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+Double Bastion is not affiliated with Telnyx or Localphone. |
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+ |
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+<span style="display:block;height:20px!important"></span> |
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+ |
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+<p align="center">Initial screen</p> |
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+<span style="display:block;height:10px!important"></span> |
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+<span style="display:block;margin:auto;width:412px;"></span> |
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+<span style="display:block;height:40px!important"></span> |
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+<p align="center">Dialpad</p> |
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+<span style="display:block;height:10px!important"></span> |
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+<span style="display:block;margin:auto;width:412px;"></span> |
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+<span style="display:block;height:40px!important"></span> |
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+<p align="center">Making calls</p> |
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+<span style="display:block;height:10px!important"></span> |
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+<span style="display:block;margin:auto;width:412px;"></span> |
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+<span style="display:block;height:40px!important"></span> |
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+<p align="center">Transferring calls</p> |
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+<span style="display:block;height:10px!important"></span> |
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+<span style="display:block;margin:auto;width:412px;"></span> |
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+<span style="display:block;height:40px!important"></span> |
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+ |
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+## Browsers |
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+<span style="display:block;height:10px!important"></span> |
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+ |
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+SIP Trip Phone works with all the major browsers. |
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+ |
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+<span style="display:block;height:20px!important"></span> |
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+ |
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+## Programming Languages |
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+<span style="display:block;height:10px!important"></span> |
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+ |
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+SIP Trip Phone only uses PHP, SQL, jQuery, CSS and HTML. This means it's robust, efficient, light-weight and easy to maintain and debug. |
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+ |
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+<span style="display:block;height:20px!important"></span> |
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+ |
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+## Minimum Requirements |
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+<span style="display:block;height:10px!important"></span> |
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+ |
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+- **Nextcloud 22+** has to be installed and properly configured, preferably by following the Install Nextcloud chapter in our guide. |
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+ |
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+- **A telnyx.com or localphone.com account and a phone number** associated with it. |
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+ |
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+- **Asterisk** (with **chan_pjsip** enabled) installed on a VPS or dedicated server. |
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+ |
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+You can also install **Coturn** as a STUN server, which helps when callers are behind routers. |
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+ |
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+<span style="display:block;height:20px!important"></span> |
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+ |
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+## Installation |
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+<span style="display:block;height:10px!important"></span> |
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+ |
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+<a href="https://www.doublebastion.com/install-nextcloud/#install-sip-trip-phone" rel="noreferrer noopener" target="_blank">This chapter</a> of our Complete Guide to a Complete Linux Server |
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+explains in detail how to install and use this application. It also contains the links to the chapters that describe how to install Asterisk and Coturn. |
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+ |
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+SIP Trip Phone is a component of RED Scarf Suite. It can be installed and used alone, but if you want to install <a href="https://www.doublebastion.com/red-scarf-suite-components/" rel="noreferrer noopener" target="_blank">all |
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+the components</a> of RED Scarf Suite, you can follow our <a href="https://www.doublebastion.com/free-server/complete-guide-to-a-complete-linux-server/" rel="noreferrer noopener" target="_blank">complete guide</a>. |
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+ |
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+<span style="display:block;height:20px!important"></span> |
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+ |
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+## Contribute |
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+<span style="display:block;height:10px!important"></span> |
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+ |
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+This is the official git repository of SIP Trip Phone. The <a href="https://github.com/DoubleBastionAdmin/sip-trip-phone" rel="noreferrer noopener" target="_blank">GitHub SIP Trip Phone |
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+repository</a> is just a pointer to this repository. We don’t use GitHub for developing SIP Trip Phone because GitHub is owned by one of the companies that proved their disrespect for |
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+digital freedom over the years and because centralized services create autonomy and privacy issues, in spite of all the benefits. |
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+ |
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+If you want to contribute code to this project, please submit <a href="https://git.doublebastion.com/sip-trip-phone/pullrequests/contrib" rel="noreferrer noopener" target="_blank">this form</a>, |
|
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+mentioning your intended changes. We'll send you the credentials needed to push code to the "contrib" branch of this repository. After we review the changes we can include them in the |
|
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+project. |
|
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+ |
|
| 107 |
+Please post any bugs that are not security related, or feature requests, on the <a href="https://git.doublebastion.com/sip-trip-phone/issues/develop" rel="noreferrer noopener" target="_blank"> |
|
| 108 |
+issue tracker</a>. If you notice bugs related to security, don’t post them on the issue tracker; instead, send them to manager [at] doublebastion [dot] com . |
|
| 109 |
+ |
|
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+<span style="display:block;height:20px!important"></span> |
|
| 111 |
+ |
|
| 112 |
+## License |
|
| 113 |
+<span style="display:block;height:10px!important"></span> |
|
| 114 |
+ |
|
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+SIP Trip Phone as a whole is licensed under the GNU Affero General Public License Version 3. If you use SIP Trip Phone or distribute it in modified or unmodified form, you will need to comply with |
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+the terms of the GNU Affero General Public License Version 3. |
|
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+ |
|
| 118 |
+This application is based on the ctxSip phone and the original copyright notice is included in the appropriate files. |
|
| 119 |
+ |
|
| 120 |
+SIP Trip Phone includes libraries licensed under different free software licenses. These libraries contain their respective original copyright notices. |