Ast 13 Beta 3 – Segfault When Calling On Pjsip Trunk With Directmedia=yes
Hello all,
I’m setting up a couple of test boxes and I’m running into a problem. What I need help with is determining whether I’m going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3
machines set up with extensions connected to each as such:
3700 —-> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to 3700 or the other way around , asterisk seg faults on both machines at roughly the same time. All connections are done using PJSIP. The crash occurs when the ringing extension is answered. If I set (directmedia=no) OR (directmedia=yes & t38_udptl=yes) on the trunk then the call completes fine. All phones and servers are on the same LAN with no firewalls active. The trunk between AST-A and AST-B is configured like this in pjsip.conf and is identical on both machines: [transport-lan] type=transport protocol=udp bind=0.0.0.0 tos
2 thoughts on - Ast 13 Beta 3 – Segfault When Calling On Pjsip Trunk With Directmedia=yes
Asterisk shouldn’t crash.
Please file a bug report ASAP at issues.asterisk.org, with a properly generated backtrace:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
Created: https://issues.asterisk.org/jira/browse/ASTERISK-24448
Let me know if you need any more information.
Thanks
-Dave