Segmentation fault when using SIP Plugin (but can receive call from a softphone)

[waldo-iton]

I have a default-setup freeswitch and tried calling default sample number and experienced the error below.

[3544671760] No stream, wait a bit in case this trickle got here before the SDP...
[3544671760] There's a message for JANUS SIP plugin
[3544671760] Creating ICE agent (controlled mode)
[3544671760] WebRTC resources freed
No WebRTC media anymore
[3544671760] No stream, wait a bit in case this trickle got here before the SDP...
[3544671760] No stream, wait a bit in case this trickle got here before the SDP...
[3544671760] No stream, wait a bit in case this trickle got here before the SDP...
[3544671760] No stream, wait a bit in case this trickle got here before the SDP...
[3544671760] No stream, wait a bit in case this trickle got here before the SDP...
Segmentation fault (core dumped)

I do not know if I was able to register successfully via Janus as I got this error message

outbound(0x7ffd800016a0): FAILED to validate sip:1000@192.168.0.41:47157;transport=udp
outbound(0x7ffd800016a0): FAILED with 481 Call/Transaction Does Not Exist

I used a softphone(xlite) registered to the same freeswitch and was able to call the account I registered via Janus SIP plugin. Calling in a different direction (browser->Janus->SIP Plugin->Freeswitch->xlite) caused the same segmentation error above.

[lminiero]

Hi, sorry for this awfully late reply.

The log you pasted doesn't contain any useful information, there may me ton of different causes to that. Please create a log with debug 5 and post it somewhere (e.g., pastebin) for us to see.

If you're familiar with gdb, please post a backtrace as well.

[lminiero]

Closing for lack of feedback.