[asterisk-bugs] [Asterisk 0017960]: [patch] SIP peer wrong URI an to: tag

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Sep 15 08:05:54 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17960 
====================================================================== 
Reported By:                adriavidal
Assigned To:                mnicholson
====================================================================== 
Project:                    Asterisk
Issue ID:                   17960
Category:                   Channels/chan_sip/General
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.8.0-beta4 
JIRA:                       SWP-2172 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-09-07 10:25 CDT
Last Modified:              2010-09-15 08:05 CDT
====================================================================== 
Summary:                    [patch] SIP peer wrong URI an to: tag
Description: 
When doing an outgoing call with a defined peer the call have wrong to:
URI
Asterisk is putting the IP of the host and not the hostname into the URI.



U 2010/09/07 17:08:47.769499 87.98.XX.XX:5060 -> 87.94.XX.XX:5060
INVITE sip:607XXXXXX at 87.94.XX.XX SIP/2.0?
Via: SIP/2.0/UDP 87.98.XX.XX:5060;branch=z9hG4bK0982bfd1?
Max-Forwards: 70?
From: "me" <sip:adriavidal at sip.proxy.net>;tag=as4582f1da?
To: <sip:607XXXXXX at 87.94.XX.XX>?
Contact: <sip:adriavidal at 87.98.XX.XX:5060>?
Call-ID: 2c210e6932761d2775058f7238c0f3ba at sip.proxy.net?
CSeq: 102 INVITE?
User-Agent: Asterisk PBX 1.8.0-beta4?
Date: Tue, 07 Sep 2010 14:46:52 GMT?
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH?
Supported: replaces, timer?
Content-Type: application/sdp?
Content-Length: 309?
?
v=0?
o=root 95065959 95065959 IN IP4 87.98.XX.XX?
s=Asterisk PBX 1.8.0-beta4?
c=IN IP4 87.98.XX.XX?
t=0 0?
m=audio 15774 RTP/AVP 18 3 101?
a=rtpmap:18 G729/8000?
a=fmtp:18 annexb=no?
a=rtpmap:3 GSM/8000?
a=rtpmap:101 telephone-event/8000?
a=fmtp:101 0-16?
a=silenceSupp:off - - - -?
a=ptime:20?
a=sendrecv?

#
U 2010/09/07 17:08:47.783243 87.94.XX.XX:5060 -> 87.98.XX.XX:5060
SIP/2.0 403 Outbound Not Allowed?
Via: SIP/2.0/UDP 87.98.XX.XX:5060;branch=z9hG4bK0982bfd1;rport=5060?
From: "me" <sip:adriavidal at sip.proxy.net>;tag=as4582f1da?
To:
<sip:607XXXXXX at 87.94.XX.XX>;tag=1ce00265432104f70edaf1386bb937d7.61ac?
Call-ID: 2c210e6932761d2775058f7238c0f3ba at sip.proxy.net?
CSeq: 102 INVITE?
Server: OpenSER (1.3.2-tls (i386/linux))?
Content-Length: 0?


that's the peer definition

[ilimit]
type=peer
host=sip.proxy.net
fromuser=adriavidal 
defaultuser=adriavidal
fromdomain=sip.proxy.net
secret=xxxxxxxxxxx
context=from-proxy
insecure=port,invite
canreinvite=no
disallow=all
allow=g729
allow=gsm
allow=g722
;allow=g723
qualify=no

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
duplicate of        0017676 [patch] host not used in invite message...
====================================================================== 

---------------------------------------------------------------------- 
 (0126985) svnbot (reporter) - 2010-09-15 08:05
 https://issues.asterisk.org/view.php?id=17960#c126985 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 286868

U   branches/1.8/channels/chan_sip.c

------------------------------------------------------------------------
r286868 | mnicholson | 2010-09-15 08:05:53 -0500 (Wed, 15 Sep 2010) | 16
lines

Set tohost to the domain specified in the configuration file instead of
the IP address of the host we are calling.

This fixes a regression introduced in r274783.

(closes issue https://issues.asterisk.org/view.php?id=17960)
Reported by: adriavidal
Patches:
      sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mich, mnicholson, adriavidal

(closes issue https://issues.asterisk.org/view.php?id=17676)
Reported by: outcast
Patches:
      sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson

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

http://svn.digium.com/view/asterisk?view=rev&revision=286868 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-09-15 08:05 svnbot         Checkin                                      
2010-09-15 08:05 svnbot         Note Added: 0126985                          
======================================================================




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