has PJ fixed the issues they had with BLF caller ID ? I had tried it a year or so ago and the NOTIFY messages for BLF were not sending the calling party... I really need to be making the move in our next production version to PJ since chan_sip is fully on its way out now..
havent tried compiling it on my RPI 5 yet to see if PJ will compile or not. Chan_sip does without issue. On Sunday, June 16, 2024 at 05:40:06 PM EDT, Michael Knill <michael.kn...@ipcsolutions.com.au> wrote: Thanks David Due to move to PJSIP in our next major release. Regards Michael Knill From: David Kerr <da...@kerr.net> Sent: Thursday, 13 June 2024 11:31 PM To: AstLinux Users Mailing List <astlinux-users@lists.sourceforge.net> Subject: Re: [Astlinux-users] PJSIP Thanks for the pointer to the custom asterisk commands, for some reason my eyes didn't pick up on those. So all is good, I used the show registrations and contacts commands. Michael, I've been putting off moving to PJSIP for so long! I'm still on Asterisk 16. But I decided I should move up to Asterisk 20 and before doing that would make the shift to PJSIP (as old SIP is officially deprecated). So I've made the move to PJSIP on 16 and if all is good, will then move to 20. The process turned out to be easier than I expected. There is a python script in the Asterisk source tree that helps a lot. It's not perfect, but it goes a long way towards creating a working pjsip.conf file out of an existing sip.conf file. It includes a commented-out section at the top which lists things it could not migrate, somewhat surprisingly for me, that included "username" fields that need to go into the authentication sections... I had to correct those manually. And it did not convert "fullname" fields which I think need to go into a callerid field, not done that yet. And then in extensions.conf you have to replace all Dial() destinations that are "SIP/<number>" with "PJSIP/<number>" and if you have syntax that looks like "SIP/<trunk>/<number>" then those need to change to "PJSIP/<number>@<trunk>" And the last thing to note is don't have both SIP and PJSIP at the same time... at least not using both your old sip.conf and new pjsip.conf files. Either remove/rename your old sip.conf or do what I did and add a noload statement to modules.conf for chan_sip. I still have to test all my esoteric paths in extensions.conf, but basic ingoing and outgoing calls are working for me. David. On Wed, Jun 12, 2024 at 8:52 PM Michael Knill <michael.kn...@ipcsolutions.com.au> wrote: Yes Im going to need to go down this path at some stage but Im not looking forward to it 🙁 Regards Michael Knill From: Home <d...@ryson.org> Sent: Thursday, 13 June 2024 8:27 AM To: AstLinux Users Mailing List <astlinux-users@lists.sourceforge.net> Subject: Re: [Astlinux-users] PJSIP For what it may be worth, I've found... pjsip show endpoints and pjsip show contacts ... to be useful. Dan -------- Original message --------From: Lonnie Abelbeck <li...@lonnie.abelbeck.com>Date: 6/12/24 6:05 PM (GMT-05:00) To: AstLinux Users Mailing List <astlinux-users@lists.sourceforge.net>Subject: Re: [Astlinux-users] PJSIP Hi David, In the Prefs -> Status Tab Options:, there are 4 pairs of these: -- Custom Asterisk Name: Custom Asterisk Command: -- Which you can label and call what commands you want. And uncheck: -- Show SIP Trunk Registrations Show SIP Peer Status -- Though I don't think there is an exact equivalent from chan_sip to chan_pjsip for status. I still use chan_sip. Lonnie > On Jun 12, 2024, at 4:17 PM, David Kerr <da...@kerr.net> wrote: > > It looks like > > pjsip list (or show) registrations > pjsip list (or show) contacts > > Gets closest to the old versions? Should the prefs panel be updated to allow > a command to be provided? > > David > > On Wed, Jun 12, 2024 at 5:10 PM David Kerr <da...@kerr.net> wrote: > I'm embarking on a long overdue conversion from SIP to PJSIP in my Asterisk > configuration. I think I have it mostly working now but I notice that in the > status page the commands to show SIP trunk and SIP peer status no longer > exist (I have noload for chan_sip.so). > > SIP Trunk Registrations:No such command 'sip show registry' (type 'core show > help sip show' for other possible commands) > > SIP Peer Status:No such command 'sip show peers' (type 'core show help sip > show' for other possible commands) > > > Are there alternative commands we could use? > > Thanks > David > > > _______________________________________________ > Astlinux-users mailing list > Astlinux-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pay...@krisk.org. _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org._______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org. _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
_______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.