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

[Asterisk Bug Tracker]

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 04:01 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
----------------------------------------------------------------------
related to          0017676 [patch] host not used in invite message...
====================================================================== 

---------------------------------------------------------------------- 
 (0126973) adriavidal (reporter) - 2010-09-15 04:01
 https://issues.asterisk.org/view.php?id=17960#c126973 
---------------------------------------------------------------------- 
The To: is fixed now, but into the from the domain is the IP from the
Asterisk himself not the domain of destination, and seems to be causing
problems with the remote proxy.

wrong capture

U 2010/09/15 10:50:13.886659 87.98.XXX.XX:5060 -> 80.94.X.XX:5060
INVITE sip:607XXXXXX at sip.proxy.net SIP/2.0?
Via: SIP/2.0/UDP 87.98.XXX.XX:5060;branch=z9hG4bK5d41ef1d?
Max-Forwards: 70?
From: "A" <sip:adriavidal at 87.98.XXX.XX>;tag=as30270354?
To: <sip:607XXXXXX at sip.proxy.net>?
Contact: <sip:adriavidal at 87.98.XXX.XX:5060>?
Call-ID: 1a50bfc674fa95b85eeedb7567a0c26f at 87.98.XXX.XX:5060?
CSeq: 102 INVITE?
User-Agent: Asterisk PBX 1.8.0-beta4?
Date: Wed, 15 Sep 2010 08:49:03 GMT?
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH?
Supported: replaces, timer?
Content-Type: application/sdp?
Content-Length: 313? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-09-15 04:01 adriavidal     Note Added: 0126973                          
======================================================================