I don't know about BLF, but I did find another change I needed to make in
my dialplan.  The CHANNEL() function has parameters that are unique to SIP
/ PJSIP.  So I was using CHANNEL(recvip) and that failed because it is
specific to SIP.  I replaced it
with CUT(CHANNEL(pjsip,remote_addr),":",1)... the CUT() is necessary
because PJSIP returns both the IP address and the port.

David.

On Sun, Jun 16, 2024 at 6:03 PM The Cadillac Kid via Astlinux-users <
astlinux-users@lists.sourceforge.net> wrote:

> 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 <
> michael.kn...@ipcsolutions.com.au> wrote:
>
>
> 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> 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>
> *Sent:* Thursday, 13 June 2024 8:27 AM
> *To:* AstLinux Users Mailing List <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>
> Date: 6/12/24 6:05 PM (GMT-05:00)
> To: AstLinux Users Mailing List <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> 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> 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
> > 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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>
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>
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> pay...@krisk.org.
>
> _______________________________________________
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> Astlinux-users@lists.sourceforge.net
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>
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> pay...@krisk.org.
> _______________________________________________
> Astlinux-users mailing list
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>
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