[asterisk-bugs] [Asterisk 0018258]: Picked call is hanged after 30 seconds (with BYE, because of timeout)

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Nov 4 19:16:05 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18258 
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Reported By:                pixall
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18258
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.0 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-11-04 19:09 CDT
Last Modified:              2010-11-04 19:16 CDT
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Summary:                    Picked call is hanged after 30 seconds (with BYE,
because of timeout)
Description: 
In our environment we have Cisco phones 7945G and 7965G, running SIP/TCP,
and one Patton PSTN (E1) gateway. Everything is in one LAN, one IP subnet,
no NAT or routing between them, no other traffic in the LAN.

If there is ingoing call to SIP phone, and other phone picks the call, the
call is sucessfully picked, but after 30 seconds it is ended - Asterisk
reports "Hanging up call XXXX - no reply to our critical packet".

It doesn't matter, if call is originated from PSTN or from another IP
phone. Also doesn't matter if it is picked using pickupgroup (dialling *8),
or if picked directly using PickUp() in dialplan, or if picked using BLF on
the phone.

Commented SIP debug is attached.
====================================================================== 

---------------------------------------------------------------------- 
 (0128641) pabelanger (manager) - 2010-11-04 19:16
 https://issues.asterisk.org/view.php?id=18258#c128641 
---------------------------------------------------------------------- 
A few things:
1. Have you read doc/sip-retransmit.txt?
2. Do not paste debug logs (see below) in your comments, upload them at
attachments.
3. Have you raise your question in the support tracker,
http://www.asterisk.org/support
4. How is this a bug?

---
We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs
from an Asterisk machine for the purpose of helping bug marshals
troubleshoot an issue:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-04 19:16 pabelanger     Note Added: 0128641                          
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