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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