I migrated from SIP to PJSIP several months ago. In general it went okay, but I ran into a problem with one DID provider... outbound would work fine, inbound calls just never arrived and nothing would show up in the asterisk debug messages. I had to tcpdump the SIP packets to figure out what was going on.
The problem was authentication on the inbound SIP requests... PJSIP/asterisk was sending back an authentication challenge on the inbound calls. This is typical for outbound, but not inbound and the DID provider could not handle. I commented out the "auth = did_provider" statement and that got it working on inbound calls. David On Sun, Jun 22, 2025 at 1:25 PM Dr. Peter Voigt <pvo...@uos.de> wrote: > Hi Lonie, > > thanks for your quick reply. Nice to hear that chan_sip support will be > continued with version 20.x giving me at least some more time for this > complex > migration. I've configured Asterisk numerous yours ago and almost left it > untoched as it is simply working. Therefore I am not an Asterisk expert at > all > and I am having only little incentive to migrate to res_pjsip with lots of > time > needed just to have things working as they to for years. > > If you ever stumble over any of these migration scripts, please let me > know to > let me give them a try. > > Regards, > Peter > > > On Sat, 2025-06-21 at 17:38 -0500, Lonnie Abelbeck wrote: > > Hi Peter, > > > > The AstLinux, Asterisk version 20.x [1] still contains chan_sip. That > will be > > the last of chan_sip support. > > > > Personally I still use chan_sip, so I have no experience migrating to > > chan_pjsip. > > > > It is my understanding there are scripts to roughly perform the > migration for > > you. > > > > Lonnie > > > > [1] https://doc.astlinux-project.org/userdoc:tt_asterisk_upgrade_version > > > > > > > > > On Jun 21, 2025, at 11:52 AM, Dr. Peter Voigt <pvo...@uos.de> wrote: > > > > > > I am currently running astlinux-1.5.10 x86_64 - Asterisk 18.26.1 using > > > chan_sip. > > > > > > If reading all information correctly, Asterisk 18 will be the last > > > Asterisk supporting chan_sip. > > > > > > So I think it is time to start migrating to res_pjsip.so. My current > > > chan_sip.so configuration is quite simple, just containing two trunk > > > definitions connecting to two different providers and two extensions > > > according > > > to my to phones. > > > > > > I have checked already that module pjsip is loaded (res_pjsip.so and > various > > > modules starting with res_pjsip). > > > > > > To get a rough understanding of the whole migration process: Will it be > > > enough > > > to migrate configuration file sip.conf to pjsip.conf and subsequently > delete > > > sip.conf. Furthermore, is it sufficient to replace any occurance of > "SIP" in > > > extensions.conf with "PJSIP"? > > > > > > I appreciate any feedback including helpful links for further reading > > > possibly > > > showing running pjsip.conf examples. > > > > > > Regards, > > > Peter > > > > > > > > > > > > _______________________________________________ > > > 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.
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