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 05:24:34Date Closed:2007-01-09 13:11:03.000-0600
Versions:Frequency of
Environment:Attachments:( 0) core1.txt
( 1) core2.txt
Description:dial plan works fine with in our production server. after i upgrade to, 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

See http://bugs.digium.com/view.php?id=7890
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, 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
sip debug
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.