Hello,
about module websocket, is it possible to associate a value with the TCP
connection and have this value readable when handling SIP requests on that
connection?
Hi Team,
I am using Jitsi client with Kamailio. I was able to IM between the clients. How can I control the Message routing in Kamailio script. Please help me with any link.
Regards,
Surya.
I made successful audio calls from browser to browser using Asterisk
13.1 and SIPML5 browser phone.
Asterisk can't manage WebRTC video calls due to lack of codec
negotiation module, but I also faced RTP ports NAT traversal issue. To
my understanding Kamailio is capable to resolve this.
Can anybody confirm that he made successful browser to browser video
calls with Kamailio sip proxy / registrar in front of Asterisk PBX.
Also, any link to good tutorial or doc pages will be appreciated.
Best Regards,
Ivan Vujisic
Hi all.
I have this setup.
Trunk--->Kamailio---->FreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
Basically this is what I'm trying to workout
FS---->kamailio---->trunk.
Any help will be much appreciated. Thanks.
AJ
