Re: Get channel variables via ARI/AMI

"TTT" <[email protected]> Mon, 26 Jun 2023 20:30:55 +0000
Newsgroups gmane.comp.telephony.pbx.asterisk.user
Message-ID <01000188f966faea-5d1c536f-fb6b-4558-a319-847fcd860144-000000@email.amazonses.com>
According to RFC3261 :

 

It is possible for the CSeq sequence number to be higher than the remote sequence number by more than one. This is not an error condition, and a UAS SHOULD be prepared to receive and process requests with CSeq values more than one higher than the previous received request.

 

So if I use a high value integer (maxint) I’m hoping the UAC will accept my message (though this will mess up future transactions, so other than BYE this is not a viable solution for an ongoing dialog).  The other problem I see is that I can't get the route set, so this this would work only with UAC’s on the same network (not NAT/proxies/etc).   If I needed to traverse NAT I would need something like Kamilio as Eric points out, to get the route set).  Or…maybe PJSIP_HEADERS will give me the route set..I need to experiment with that one).

 

Since this is just for my learning, I want to see if I can hangup a call in progress, running through the asterisk server.  This is fun for my learning, so I realize there is of little practical value :)

 

Brian

 

 

From: asterisk-users [mailto:[email protected]] On Behalf Of Joshua C. Colp
Sent: Monday, June 26, 2023 3:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]>
Subject: Re: [asterisk-users] Get channel variables via ARI/AMI

 

On Mon, Jun 26, 2023 at 4:35 PM TTT <[email protected] <mailto:[email protected]> > wrote:

I think that’s getting me close.  I’m trying to get (or recreate) the FROM and TO lines of the header, from a system running PJSIP.  I think if I use CHANNEL to get local_uri and local_tag I can recreate a FROM line like:

FROM=<URI>;tag=TAG

 

And if I use CHANNEL to get remote_uri and remote_tag I can recreate a FROM line like:

TO=<URI>;tag=TAG

 

Would it be correct to assume that with this info (and ip:port info) I should be able to send a UDP SIP message from the PBX to the UA which appears to be part of the current call dialog?  I realize this is an odd thing to do, but I’m just interested in technical feasibility at this point.  Before I try to code this I want to ensure I’m not missing something stupid.

 

Probably not. Sequence number also matters.

 

-- 

Joshua C. Colp

Asterisk Project Lead

Sangoma Technologies

Check us out at www.sangoma.com <http://www.sangoma.com>  and www.asterisk.org <http://www.asterisk.org>

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