[asterisk-bugs] [Asterisk 0013126]: SIP/2.0 400 SIP Parser Error : Missing '@', line 3, column 26

noreply at bugs.digium.com noreply at bugs.digium.com
Mon Jul 21 16:27:50 CDT 2008


The following issue requires your FEEDBACK. 
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http://bugs.digium.com/view.php?id=13126 
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Reported By:                tkfast
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13126
Category:                   Channels/chan_sip/Registration
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.21 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             07-21-2008 10:43 CDT
Last Modified:              07-21-2008 16:27 CDT
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Summary:                    SIP/2.0 400 SIP Parser Error : Missing '@', line 3,
column 26
Description: 
Linksys WIP330 Phones cause Sip Parse Errors and Crash Asterisk Completely,
It crashes most when you get a new incoming call then it will show No
Caller ID you will have silence on both ends of the line.
Then you have to restart asterisk, sometimes it will lock up asterisk
where you can't even do a stop now and you have to kill the process. When
this happens is throws errors like this.
SIP/2.0 400 SIP Parser Error : Missing '@', line 3, column 26

If you turn off the Wip330 phone the systems work great. So I know it is
directly related to how theses register with SIP.

Occurring on Multiple Server Running Different Versions of Asterisk.

Asterisk 1.4.21.1

phonesystem*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 400 SIP Parser Error : Missing '@', line 3, column 26
From: "No CallID" ;tag=as678f21e5
To:
Call-ID: 5848a4a725494e8c0bdd274e15c14ecc at 192.168.1.25
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.25:5060;rport=5060;branch=z9hG4bK7d9f6686
User-Agent: Asterisk PBX
Max-Forwards: 70
Supported: replaces
Date: Tue, 15 Jul 2008 22:16:25 GMT
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Content-Length: 0


Asterisk 1.4.20.1

[May 16 22:08:00] WARNING[14761]: chan_sip.c:6772
determine_firstline_parts: Bad request protocol SIP Parser Error : Missing
'@', line 3, column 26
[May 16 22:08:00] NOTICE[14761]: chan_sip.c:15379 handle_request: Unknown
SIP command '2070SIP/2.0' from '97.101.232.29'
-- Got SIP response 400 "SIP Parser Error : Missing '@', line 3, column
26" back from 97.101.232.29
[May 16 22:08:04] NOTICE[14761]: chan_sip.c:15766 sip_poke_noanswer: Peer
'104' is now UNREACHABLE! Last qualify: 409

I have also tried to see if it does it on the follow versions of asterisk
and it does the same thing.

asterisk-1.4.17
asterisk-1.4.18
asterisk-1.4.18.1
asterisk-1.4.19.1
asterisk-1.4.19.2
asterisk-1.4.21-rc1 

OS Used: Debian Etch
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Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-21-08 16:27  russell        Status                   new => feedback     
======================================================================




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