I’m running freepbx v14. I’m afraid to upgrade Asterrisk worried my phones become failed to register on server. Is freepbx v14 will ever stop supporting the chan_sip potocol?
chan_sip is not a protocol. It is one of two channel drivers that support the SIP protocol.
The use of chan_sip is already strongly discouraged in favour of chan_pjsip, and chan_sip has to be actively enabled if installing Asterisk from scratch. chan_sip is expected to be removed from Asterisk in, maybe two years from now, once the remaining things that only chan_sip can do have been addressed.
I have several servers running on v14 for a few years. It’ll take some effort to move them to chan_pjsip. Is v14 going away along with chan_sip?
It isn’t actually supported by chan_sip either. The difference is that chan_pjsip actively rejects it, to avoid unpredictable behaviour, whereas, with chan_sip, you use it at your own risk.
My understanding is chan_sip won’t be removed completely until the edge cases are covered by chan_pjsip, although I don’t know whether TEL: URI support is actually on the list.
In practice chan_sip is already unsupported, as Sangoma don’t support it and there has been almost no recent input from the open source community.
My ITSP/fixed line provider uses Huawei IMS/Sip system which uses Tel:URI. I have used Freeswitch with it in the past and it works well like the chan_sip. Hopefully, somebody is tracking this for chan_pjsip in Asterisk.
Some of my phones are old…does the extension phones have to be a pjsip compatible? Or I just have to reconfigure them to match the port number used by the server for chan_pjsip?
SIP is SIP, provided the basic user/auth/server/port conventions are followed, most things should just work.
Chan_sip is slated for complete removal in Asterisk 21. As for the community support for chan_sip, not only has it been sparse but from what I’ve seen from the devs, the community fixes for a problem/feature add usually ends up causing regression in chan_sip.