Re: [Astlinux-users] PJSIP migration

[The C. K. <eld...@ya...>]

 PJSIP is confusing at first but after a short time its flow is pretty easy... the part i miss most about CHAN_SIP is the 'sip show peers' in a nice compact easy to read format.. pjsip show endpoints is not so easy..  several lines per station / trunk.. 
you can use the wizard format to have all of your PJSIP parameters in one entry vs multiple objects per peer / trunk. im not sure if astlinux has the wizrd co,piled in or not (ive been using regular asterisk for quite some time..  
    On Sunday, June 22, 2025 at 02:03:12 PM EDT, David Kerr <da...@ke...> wrote:  
 
 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 <pv...@uo...> 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 <pv...@uo...> 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
> > 
> > 
> > 
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