|Summary:||ASTERISK-07761: Asterisk crashes in handle_request_refer() when an agent does a transfer|
|Reporter:||Peng Yong (ppyy)||Labels:|
|Date Opened:||2006-09-18 05:24:34||Date Closed:||2007-01-09 13:11:03.000-0600|
|Environment:||Attachments:||( 0) core1.txt|
( 1) core2.txt
|Description:||dial plan works fine with 18.104.22.168 in our production server. after i upgrade to 22.214.171.124, i got two crash today in 3 hours.|
asterisk crash after handle_request_refer().
the asterisk has a small queue, and crash happens after the agent transfer to a extension in remote openser.
|Comments:||By: Leonardo Gomes Figueira (sabbathbh) 2006-09-18 07:35:45|
I think it may be related.
By: Serge Vecher (serge-v) 2006-09-18 10:33:30
ppyy: is it possible to obtain a SIP debug from the console prior to the crash?
By: Peng Yong (ppyy) 2006-09-18 12:00:17
sabbathbh, we does not issue a "show channels".
serge-v, we have downgrade to 126.96.36.199, and i will try to catch the cash in a test platform later.
By: jmls (jmls) 2006-11-01 05:59:20.000-0600
ppyy, were you able to catch that crash on the test system ?
By: Olle Johansson (oej) 2006-11-07 02:37:04.000-0600
I need a SIP debug for this.
By: Serge Vecher (serge-v) 2006-11-10 11:22:48.000-0600
Please do the following:
1) Prepare test environment (reduce the amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
console => notice,warning,error,debug
3) restart Asterik.
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
5) Save complete console log to file and _attach_ said file to the bug.
i.e. asterisk -Tvvvvvdddddgc | tee /tmp/sipdebug.txt
By: Olle Johansson (oej) 2006-11-12 15:14:52.000-0600
I do need more information, since the two core files are very different. Thanks.
By: Olle Johansson (oej) 2006-12-01 03:56:45.000-0600
Can you please return with more information, as well as test with latest 1.2 from svn. There's been a lot of changes to chan_sip lately. Thanks.