[asterisk-users] Getvar of CHANNEL not working for a couple of items

TTT lists at telium.io
Thu Jul 6 11:23:06 CDT 2023


I found a clue as to why the second leg is not returning a local or remote address:

 

[2023-07-06 11:40:35] WARNING[253072]: pjsip/dialplan_functions.c:903 channel_read_pjsip: No transport information for channel PJSIP/222-0000007d

[2023-07-06 11:40:35] WARNING[935126]: func_channel.c:527 func_channel_read: Unknown or unavailable item requested: 'pjsip,local_addr'

[2023-07-06 11:40:35] WARNING[40100]: pjsip/dialplan_functions.c:917 channel_read_pjsip: No transport information for channel PJSIP/222-0000007d

[2023-07-06 11:40:35] WARNING[935126]: func_channel.c:527 func_channel_read: Unknown or unavailable item requested: 'pjsip,remote_addr'

 

Well…maybe not a root cause but certainly something of interest.  This works with the first leg, but not the second leg of the call.

 

From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp
Sent: Wednesday, July 5, 2023 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Getvar of CHANNEL not working for a couple of items

 

On Wed, Jul 5, 2023 at 12:50 PM TTT <lists at telium.io <mailto:lists at telium.io> > wrote:

  Channel A: "1688509741.112" , name:  "PJSIP/111-00000064" , is originator:  Y , call-Id:  "u.l6kcou25cax60 at mydomain.com <mailto:u.l6kcou25cax60 at mydomain.com> " , local_uri:  "<sip:222 at mydomain.com <mailto:sip%3A222 at mydomain.com> ;user=phone>" , local_tag:  "1734d973-c4da-4ae8-a37d-5f7065f1fe54" , local_addr:  "172.31.253.4:5060 <http://172.31.253.4:5060> " , remote_uri:  "\\\"TestPhone x111\\\" <sip:111 at mydomain.com <mailto:sip%3A111 at mydomain.com> >" , remote_tag:  "yinue4v5ufa4" , remote_addr:  "172.31.253.20:5060 <http://172.31.253.20:5060> "

 

 

  Channel B: "1688509741.113" , name:  "PJSIP/222-00000065" , is originator:  N , call-Id:  "1f104544-fc1a-4414-ba74-68c526e294de" , local_uri:  "\\\"TestPhone\\\" <sip:111 at 172.31.253.4 <mailto:sip%3A111 at 172.31.253.4> >" , local_tag:  "ac5eeb59-f559-4bb7-a3c2-170ca7f05f8b" , local_addr:  "" , remote_uri:  "<sip:222 at 172.31.253.20 <mailto:sip%3A222 at 172.31.253.20> ;line=46922>" , remote_tag:  "klwqxe1fvt5wk" , remote_addr:  ""

 

And here's what seems strange:

Channel A's local_uri looks like Channel B's uri

Channel A's remote_uri looks like channel A's uri

Channel B's local_uri looks like channel A's uri

Channel B's remote_uri looks like channel B;s uri

 

These aren't strange. They look alike because of callerid and target dialed information. They are still independent call legs.

 

 

I’m having trouble understanding your explanation (googling just led me to generic callerid and target info).  I thought a phone’s local_uri would be how to reach that phone (not the other party), and vice versa for the remote_uri.  If the above URI info is correct then I must misunderstand their meaning.  Could you provide more explanation on how to interpret them (why they seems reversed to me), or a link?

 

I assumed the remote & local URI where equivalent to the to & from lines (respectively) in the invite…

 

They are the From and To header, but what remote_uri and local_uri refers to changes depending on the direction of the SIP dialog.

 

Received call: From = remote_uri, To = local_uri

Sent call: From = local_uri, To = remote_uri

 

The contents of each depend on callerid information, settings, the Contact of the target when doing an outgoing call, what the remote endpoint chose for To URI on a received call.

 

-- 

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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