FreeSwitch WebRTC Call Termination

Asked Oct 14 '16 at 04:16

Active Oct 14 '16 at 18:07

Viewed 341 times

I have FreeSwitch working with SIP Clients for

I have configured WebRTC with SIPML5 clients and it is working on following scenarios

for Extension to PSTN/ Gateway FreeSwitch routes call to TRUNK but its not connected.

How can I get it working ?? What is missing parameter?? I think something at bridge is required to

FS Console logs are available here

asked Oct 14 '16 at 04:16

For Guru

Just capture the sip log from invite to hangup and paste in pastebin. This info is not enough – Sasi Varunan Oct 18 '16 at 16:58
paste the freeswitch log with more detail in pastebin – suren Oct 20 '16 at 08:13

0 Answers0