Summary:	ASTERISK-16004: Call transfer from Voicemail or Queue Application result Asterisk to crash. (SIP REFER)
Reporter:	Alexander Nazaruk (anazaruk)	Labels:
Date Opened:	2010-04-22 22:23:12	Date Closed:	2011-06-07 14:01:06
Priority:	Critical	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/Transfers
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) gdb.txt ( 1) thread_.txt
Description:	Polycom and Linksys SIP phones result same crash. 1. Dial voicemail or queue. 2. While checking voicemail or being in queue, put call on hold and dial another extension from the same IP phone. 3. Pick up the phone on another extension you dial. 4. Push transfer button. IP phone will send SIP REFER message to asterisk witch will result asterisk to CORE. .. It only crashes when you trying to transfer Voicemail or Queue to Extension. While trying to transfer extension to Queue or Voicemail it will not crash. Other call transfers work fine. (ext to ext or ext to PSTN number out SIP) ****** ADDITIONAL INFORMATION ****** Removed inline backtrace - pabelanger
Comments:	By: Paul Belanger (pabelanger) 2010-04-22 23:05:19 Your backtraces are optimize out (see below) --- Thank you for your bug report. In order to move your issue forward, we require a backtrace from the core file produced after the crash. Please see the doc/backtrace.txt file in your Asterisk source directory. Also, be sure you have DONT_OPTIMIZE enabled in menuselect within the Compiler Flags section, then: make install after enabling, reproduce the crash, and then execute the instructions in doc/backtrace.txt. When complete, attach that file to this issue report. Thanks! By: Alexander Nazaruk (anazaruk) 2010-04-23 02:57:18 please close this bug. I have figured it out. Thank you By: Paul Belanger (pabelanger) 2010-04-23 08:08:36 Closed per request.