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
<[email protected]> wrote:
Thanks David
Due to move to PJSIP in our next major release.
Regards
Michael Knill
From: David Kerr <[email protected]>
Sent: Thursday, 13 June 2024 11:31 PM
To: AstLinux Users Mailing List <[email protected]>
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
<[email protected]> 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 <[email protected]>
Sent: Thursday, 13 June 2024 8:27 AM
To: AstLinux Users Mailing List <[email protected]>
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
<[email protected]>Date: 6/12/24 6:05 PM (GMT-05:00) To: AstLinux Users
Mailing List <[email protected]>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 <[email protected]> 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 <[email protected]> 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
>
>
> _______________________________________________
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> [email protected]
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>
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> [email protected].
_______________________________________________
Astlinux-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
[email protected]._______________________________________________
Astlinux-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
[email protected].
_______________________________________________
Astlinux-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
[email protected]. _______________________________________________
Astlinux-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to
[email protected].
