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Summary:ASTERISK-05244: [chan_sip/app_dial?] CANCEL sent after 30 seconds when timeout >30
Reporter:jaredmauch (jaredmauch)Labels:
Date Opened:2005-10-05 08:56:44Date Closed:2011-06-07 14:10:42
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Applications/app_dial
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I have IOS gateways doing SIP for my PSTN linkup.  When sending calls to them, eg:

Dial(SIP/1734913nnnn@pstnlink) or

Dial(SIP/1734913nnnn@pstnlink|60)

They receive a SIP CANCEL from Aasterisk, I've tried to find the problem in the source a few times but have not had much luck.  This is mostly a query for some help on this, but also to see if anyone else has had Dial/SIP not honor the 30s timeout as i'm seeing this on more than one asterisk cvs head box.

****** ADDITIONAL INFORMATION ******

Here's what's received by the IOS Gateway about 30 seconds after it sends a 183.

Sep  7 12:22:36.842 EDT: Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK764fdcd7;rport
From: "Jared Mauch" <sip:104@X.X.X.X>;tag=as79ab38ec
To: <sip:1734913nnnn@X.X.X.X>;tag=D033D00A-FA5
Date: Wed, 07 Sep 2005 16:22:34 GMT  
Call-ID: 4297863a69ed6bfd624ce55f2666f264@X.X.X.X
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:17349138660@X.X.X.X:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 238

v=0
o=CiscoSystemsSIP-GW-UserAgent 9746 7705 IN IP4 X.X.X.X
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=audio 17778 RTP/AVP 0 101
c=IN IP4 X.X.X.X
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16



Sep  7 12:23:04.901 EDT: Received:
CANCEL sip:1734913nnnn@X.X.X.X SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK764fdcd7;rport
From: "Jared Mauch" <sip:104@X.X.X.X>;tag=as79ab38ec
To: <sip:1734913nnnn@X.X.X.X>
Contact: <sip:104@X.X.X.X>
Call-ID: 4297863a69ed6bfd624ce55f2666f264@X.X.X.X
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
Comments:By: Olle Johansson (oej) 2005-10-05 09:18:33

Please read the bug guidelines. We need a full SIP debug, not just a few packets. Turn on verbose to 4, debug to 4 and turn on SIP debug. Make sure you capture debug output to console in logger.conf.

Also need to see "sip show settings". Thank you for a quick reply!
By: jaredmauch (jaredmauch) 2005-10-05 09:37:21

sigh, i see my issue, go ahead and close this.
By: Olle Johansson (oej) 2005-10-05 09:46:07

Please do *NOT* add debug to the bug tracker, attach it as a file. And you forgot all about the debug and verbose. Please try again.
By: Olle Johansson (oej) 2005-10-05 09:51:33

---Closed on reporters request.
Thank you for reporting what you once thought was a bug :-)
It happens to everyone.

/Olle