[Home]

Summary:ASTERISK-05259: chan_sip doesn't form the SIP URI correctly when placing outgoing SIP calls received from another * box with IAX
Reporter:lancey (lancey)Labels:
Date Opened:2005-10-06 19:57:32Date Closed:2011-06-07 14:10:13
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:If there's no "fromuser" in sip.conf, the URI should get formed using the CallerID. But when the call is received from another * box, it gets set to "Unknown".

****** ADDITIONAL INFORMATION ******

Here's my scenarion:
* box ---(IAX2)---> * box ----(SIP)----> some.sip.peer

Some output from the middle * box:

   -- Accepting AUTHENTICATED call from the.first.asterisk.box
   -- Executing Set("IAX2/the.first.asterisk.box-3", "CALLERID(number)=0123456") in new stack
   -- Executing Verbose("IAX2/the.first.asterisk.box-3", "0123456") in new stack (this is indeed Verbose(${CALLERID}) to make sure it is set ok)
   -- Executing Dial("IAX2/the.first.asterisk.box-3", "SIP/some.sip.peer/00123456789") in new stack
We're at the.middle.asterisk.box port 47326
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
12 headers, 10 lines
Reliably Transmitting (no NAT) to some.sip.peer:5060:
INVITE sip:00123456789@some.sip.peer SIP/2.0
Via: SIP/2.0/UDP the.middle.asterisk.box:5060;branch=z9hG4bK42f083f2;rport
From: "Unknown" <sip:Unknown@the.middle.asterisk.box>;tag=as098e3719
To: <sip:00123456789@some.sip.peer>
Contact: <sip:Unknown@the.middle.asterisk.box>

We get "Unknown@" despite the caller ID is set...

Here's a second scenario:
IP phone ----(IAX2)---> * box ----(SIP)---> some.sip.peer

Output from the same middle * box:

From: "0123456" <sip:0123456@the.middle.asterisk.box>;tag=as22025fe1
To: <sip:00123456789@some.sip.peer>;tag=62ff4c0069ff4510ff000011ffff7aff
Contact: <sip:0123456@the.middle.asterisk.box>

Everything flows OK.

If there's some more output you need, I'm keeping an eye on this bug.
Comments:By: lancey (lancey) 2005-10-07 05:13:40

There could be something in relation with bug ASTERISK-518225.
By: lancey (lancey) 2005-10-10 19:40:47

Putting SetCallerPres(allowed) before dialing out solves the problem. Now the question is - how is default setting of Caller Presentation worked out?
By: Olle Johansson (oej) 2005-10-12 00:09:35

Replicating this and adding caller ID pres to dumpchan, we see that the IAX call changes caller ID pres to "Number unavailable", so this is *not* a SIP bug, but an IAX bug. Happily changing category and forgetting all about this bug report...
By: Mark Spencer (markster) 2005-10-12 01:18:01

Fixed in CVS head.
By: lancey (lancey) 2005-10-12 01:40:36

I'm now retesting, but, anyways, how is the default Caller Presentation set?
By: Digium Subversion (svnbot) 2008-01-15 15:50:33.000-0600

Repository: asterisk
Revision: 6748

U   trunk/channels/chan_iax2.c

------------------------------------------------------------------------
r6748 | markster | 2008-01-15 15:50:33 -0600 (Tue, 15 Jan 2008) | 2 lines

Don't override calling presentation if *name* is present (bug ASTERISK-5259)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=6748