[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 18:26:30 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: file
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Project: Asterisk
Issue ID: 11939
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: major
Priority: normal
Status: assigned
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:
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Date Submitted: 02-06-2008 13:07 CST
Last Modified: 02-06-2008 18:26 CST
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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
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Relationships ID Summary
----------------------------------------------------------------------
duplicate of 0011916 Asterisk don't get the BYE packet from ...
related to 0011938 Invalid interpretation of INVITE SIP frame
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falves11 - 02-06-08 18:26
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I did a painstaking regression analysis and the last version that works is
99082, when you jump to 99085, the contact field is malformed and asterisk
can not talk to many sip end-points. I think that I am the only one in the
world that is using Trunk for business on a high density wholesale model,
otherwise, somebody else must have found out about this issue. The change
happenned in the jump from 99082 to 99085. Since we are now in 102726+, I
think that we should stop here and fix this issue. I had to go back and use
a previous version, but I want to use the latest memory fixes. Please help.
Issue History
Date Modified Username Field Change
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02-06-08 18:26 falves11 Note Added: 0081825
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