Diagnosing Asterisk Segmentation Fault on STM32MP157f-DK2 When Connecting via Mizudroid SIP Client

Hello, I’m working on a project using an STM32MP157f-DK2 board running an Asterisk server on a Bootlin Linux image. The board is connected to my home network via a D-Link router configured as a network switch (DHCP disabled). Initially, the server crashed with a segmentation fault immediately after starting. After modifying the pjsip.conf file, the server now starts successfully. However, it crashes with a segmentation fault when I try to connect from my phone using the Mizudroid SIP client app.
Could this issue be related to my router setup, Asterisk configuration, or Bootlin environment? Any suggestions on diagnosing and resolving this segmentation fault would be greatly appreciated.
4 Replies
Ming
Ming5d ago
Some of my fist thoughts: When playing with Asterisk, I had no problems with a DHCP setup Assuming that the Asterisk server is now starting OK. Is it stable - as in does it all stay up without a segfault without any clients connecting to it? Have you tried just starting it and leaving it for an hour or so? If you think the client is causing issues, have you tried a different client? I used LinPhone ( https://www.linphone.org/en/homepage-linphone/ ) Another thing to try if you have a PC available would be to run the Asterisk server on that to see if it is a resource issue. When I did it, I used a PC running Debian, but then I was trying to integrate FXO/FXS adapters into my solution.
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R1Tap
R1Tap5d ago
👍🏼 will try once again by reseting my router, Yes the server is being active when kept idle. Yes. I've used linphone too I have used different conf file while testing asterisk on my arch-laptop ( since i have enabled asterisk in buildroot packages i thought the code would be different ) and it was working perfectly well i connected 2 devices with linphone When i used the same pjsip.conf file as of my laptop's asterisk, here i was getting segfault just after starting the server
Ming
Ming5d ago
Another thing to try may be falling back to using chan_sip instead of chan_pjsip. The old sip may be unsupported, but that would tell you if the problem is with pjsip code/config or the asterisk code.
Ming
Ming5d ago
@R1Tap Your config files look fine - and as they work on your main server, they should therefore work! A few questions: - What version of Asterisk are you running? - Are you running in verbose mode to see what it is doing before it crashes? asterisk -cvvvvv Also, have you looked at the PJSIP troubleshooting guide? https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Asterisk-PJSIP-Troubleshooting-Guide/ Another thing, are you getting core dumps from the crash? There is a guide on collecting debug information at https://docs.asterisk.org/Operation/Logging/Collecting-Debug-Information/ This may help in giving a hint as to what is happening.
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