Ast 13 Beta 3 – Segfault When Calling On Pjsip Trunk With Directmedia=yes

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Asterisk Users 2 Comments

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

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