[asterisk-bugs] [Asterisk 0011939]: The "contact" section of the invite is wrong and many gateways reject the invite

noreply at bugs.digium.com noreply at bugs.digium.com
Wed Feb 6 13:12:37 CST 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11939 
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Reported By:                falves11
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11939
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 102726 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             02-06-2008 13:07 CST
Last Modified:              02-06-2008 13:12 CST
====================================================================== 
Summary:                    The "contact" section of the invite is wrong and
many gateways reject the invite
Description: 
If you look at this invite, the contact section has a port=0
"Contact: <sip:7864335989 at 67.110.179.252:0>", and that causes the call to
fail with error SIP/2.0 400 Invalid Contact
It started with the latest version, it did not happen before.

INVITE sip:18183868429 at 67.203.64.22 SIP/2.0
Via: SIP/2.0/UDP 67.110.179.252:5060;branch=z9hG4bK4703c6d5;rport
Max-Forwards: 70
From: "7864335989" <sip:7864335989 at minixel.com>;tag=as5d7e4f92
To: <sip:18183868429 at 67.203.64.22>
Contact: <sip:7864335989 at 67.110.179.252:0>
Call-ID: 17eccf585f82e1ef7f4017bb06dad2ec at minixel.com
CSeq: 102 INVITE
User-Agent: AsteriskSystemsSIP-GW-UserAgent
Date: Wed, 06 Feb 2008 19:02:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 361

Call-ID: 17eccf585f82e1ef7f4017bb06dad2ec at minixel.com
CSeq: 102 INVITE
User-Agent: AsteriskSystemsSIP-GW-UserAgent
Date: Wed, 06 Feb 2008 19:02:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 361

v=0
o=root 235140376 235140376 IN IP4 67.110.179.252
s=Asterisk
c=IN IP4 67.110.179.252
t=0 0
m=audio 30630 RTP/AVP 18 4 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 18183868429 at 67.203.64.22
Sipserver*CLI>
<--- SIP read from UDP://67.203.64.22:5060 --->
SIP/2.0 400 Invalid Contact
Via: SIP/2.0/UDP
67.110.179.252:5060;received=67.110.179.252;branch=z9hG4bK4703c6d5;rport=5060
From: "7864335989" <sip:7864335989 at minixel.com>;tag=as5d7e4f92
To: <sip:18183868429 at 67.203.64.22>;tag=aprqngfrt-nk2e7630000c6
Call-ID: 17eccf585f82e1ef7f4017bb06dad2ec at minixel.com
CSeq: 102 INVITE

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0011938 Invalid interpretation of INVITE SIP frame
====================================================================== 

---------------------------------------------------------------------- 
 putnopvut - 02-06-08 13:12  
---------------------------------------------------------------------- 
This is the same issue as reported in 11938. Do you have "insecure=very" in
your sip.conf? 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
02-06-08 13:12  putnopvut      Note Added: 0081806                          
======================================================================




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