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<mailto: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<mailto:d...@ryson.org>>
Sent: Thursday, 13 June 2024 8:27 AM
To: AstLinux Users Mailing List 
<astlinux-users@lists.sourceforge.net<mailto: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<mailto:li...@lonnie.abelbeck.com>>
Date: 6/12/24 6:05 PM (GMT-05:00)
To: AstLinux Users Mailing List 
<astlinux-users@lists.sourceforge.net<mailto: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<mailto: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<mailto: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<mailto:Astlinux-users@lists.sourceforge.net>
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>
> Donations to support AstLinux are graciously accepted via PayPal to 
> pay...@krisk.org<mailto:pay...@krisk.org>.




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