Hello, I'd like share with you my tests about PJSIP channel with the aim of improving the functioning of the channel: * Multi domain support not work correctly: https://issues.asterisk.org/jira/browse/ASTERISK-26026 * Different context subscribe for each endpoint not possible: https://issues.asterisk.org/jira/browse/ASTERISK-25471 * BLF don't work correctly on my tests with X-Lite, BRIA, JiTSI. Only work partially with microsip but because this softphone use the same SIP STACK (PJSIP). I test BLF with the latest Asterisk version and latest FreePBX version. The problem is when a softphone is on the phone on the other softphone appear off-line. On Jitsi not is possible know if a Endpoints is or not online The main idea of the new channel was working on a multi-domain environment, have more then one device registered with same credentials and have more stability. Be Better still with Asterisk 1.11.X? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160717/34b1f1ed/attachment.html>
Annus Fictus wrote: <snip>> * Different context subscribe for each endpoint not possible: > https://issues.asterisk.org/jira/browse/ASTERISK-25471This was actually taken through the process by someone else recently[1] and will be in 13.11. [1] https://gerrit.asterisk.org/#/c/3145/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
2016-07-17 14:30 GMT+02:00 Annus Fictus <annusfictus at gmail.com>:> The main idea of the new channel was working on a multi-domain environment >For now, to my experience, it's more future-proof compliant to use a prefix in the SIP username than multi-domain environment. Even if the multi-domain support was perfect in Asterisk, we tested some crappy SIP endpoints where in fact, even if you configure a domain name everywhere in the configuration, you have only IPs in SIP packets. We have that on production for our cloud plateform, it works pretty well and also simplify whitelabel handling. Moreover, if you have a good provisioning support, it will be invisible for your users.>When I see the time needed to really use on production the SNI feature in SSL, and you have only 5 majors HTTP endpoints (aka Web browsers). In the SIP world, I'm not sure you can use multi domain except if you can force the SIP endpoints used by your clients.> , have more then one device registered with same credentials and have more > stability. >Since 13.9.1, we have a better experience of pjsip. Nevertheless, not yet massively used on production for now, we planned to migrate endpoint by endpoint to minimize the risk.> Be Better still with Asterisk 1.11.X? >Maybe you could use Asterisk 13 with chan_sip to start, it works pretty well and already think to support chan_pjsip in the same time. The benefit to think about that if one day you need to use an alternative channel like chan_iax2, it should be easier to implement for you.> Regards > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160717/f40cd7d7/attachment.html>
I agree the multi-domain environment is a nice idea, but too many endpoints don't properly support. We to use a prefix in the SIP username for multi-domain environments. Thanks Bryant ---------------------------------------- From: "Ludovic Gasc" <gmludo at gmail.com> Sent: Sunday, July 17, 2016 5:20 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] PJSIP - State of the art 2016-07-17 14:30 GMT+02:00 Annus Fictus <annusfictus at gmail.com>: The main idea of the new channel was working on a multi-domain environment For now, to my experience, it's more future-proof compliant to use a prefix in the SIP username than multi-domain environment. Even if the multi-domain support was perfect in Asterisk, we tested some crappy SIP endpoints where in fact, even if you configure a domain name everywhere in the configuration, you have only IPs in SIP packets. We have that on production for our cloud plateform, it works pretty well and also simplify whitelabel handling. Moreover, if you have a good provisioning support, it will be invisible for your users. When I see the time needed to really use on production the SNI feature in SSL, and you have only 5 majors HTTP endpoints (aka Web browsers). In the SIP world, I'm not sure you can use multi domain except if you can force the SIP endpoints used by your clients. , have more then one device registered with same credentials and have more stability. Since 13.9.1, we have a better experience of pjsip. Nevertheless, not yet massively used on production for now, we planned to migrate endpoint by endpoint to minimize the risk. Be Better still with Asterisk 1.11.X? Maybe you could use Asterisk 13 with chan_sip to start, it works pretty well and already think to support chan_pjsip in the same time. The benefit to think about that if one day you need to use an alternative channel like chan_iax2, it should be easier to implement for you. Regards -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160718/2b3b31f8/attachment.html>