Summary:	ASTERISK-26035: PJSIP driver, TLS client in ringgroup crashes asterisk with incoming trunk call
Reporter:	J. Lance Cotton (lcotton)	Labels:
Date Opened:	2016-05-18 10:16:06	Date Closed:	2016-05-18 12:33:45
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	pjproject/pjsip
Versions:	13.8.2 13.9.0 13.9.1	Frequency of Occurrence	Constant
Related Issues:
Environment:	FreePBX Distro 10.13.66-11. Mix of hardware clients (Panasonic KX-TG550, Snom M9) and softphone clients (MicroSIP 3.12.1, win32)	Attachments:	( 0) backtrace_full.txt ( 1) backtrace.txt ( 2) bt_threadtrace.txt ( 3) PJSIP_Log_withTLS_Crash.txt ( 4) TLS_Crash.txt ( 5) UDP_NoCrash.txt
Description:	Have 6 extensions in a ringgroup. One is a MicroSIP 13.12.1 softphone. The other 5 are hardware IP (SIP) devices that do not support TLS, so they connect by UDP all the time. When the softphone is connected by TLS transport and a call comes in from our SIP trunk provider the call is directed to the ringgroup. Asterisk crashes. When the softphone is set to connect to the UDP transport, no crash. When the softphone is using the TLS transport and a call comes in from the trunk provider, but doesn't go through a ringgroup, no crash. When the ringgroup is dialed directly from another extension in the office, no crash (TLS). PJSIP Log of crash, File: TLS_Crash.txt Same setup, but softphone using UDP transport (only change), File: UDP_NoCrash.txt
Comments:	By: Asterisk Team (asteriskteam) 2016-05-18 10:16:06.806-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Joshua C. Colp (jcolp) 2016-05-18 10:19:59.558-0500 Thank you for the crash report. However, we need more information to investigate the crash. Please provide: 1. A backtrace generated from a core dump using the instructions provided on the Asterisk wiki [1]. 2. Specific steps taken that lead to the crash. 3. All configuration information necesary to reproduce the crash. Thanks! [1]: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace By: J. Lance Cotton (lcotton) 2016-05-18 10:40:57.080-0500 The 3 different outputs from gdb requested in the WIKI docs for getting a Backtrace. By: Joshua C. Colp (jcolp) 2016-05-18 10:47:31.172-0500 You are running a PJSIP which has assertions enabled, these can cause aborts when normal circumstances are encountered and are used for development purposes. Disabling assertions can be done by building PJSIP with NDEBUG defined. This is done automatically if using the bundled version according to the wiki[1]. The above assertion is happening because the transport is "LS" on the Contact instead of "TLS". Please provide the PJSIP debug (pjsip set logger on) showing the SIP traffic involving the TLS device - if registering then the REGISTER and such. [1] https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled By: J. Lance Cotton (lcotton) 2016-05-18 10:54:23.275-0500 PJSIP log for this client in TLS, from REGISTER until crash. By: J. Lance Cotton (lcotton) 2016-05-18 10:56:49.391-0500 Note this is FreePBX distro, so I am not prepared to build PJSIP on my own. By: J. Lance Cotton (lcotton) 2016-05-18 12:28:07.845-0500 Further investigation shows that the output of the script dialparties.agi is the culprit, which is a FreePBX issue. This issue can be closed.