Hello,
There is no proper signalling when call is made between extension to extension,
since phones are in natted environment and server is on public ip, it donot works even if
i have nat = (force_rport/yes/comedia)
Regards
Sam
When exten to exten call is established , when nat = yes it connects properly with rtp but when disconnected
the both legs do not disconnects leading to 481 transaction does not exist.
Regards
Sam
this was working good on 1.4.39 but here the signalling is not proper between exten to exten.
is there any parameter i am missing on 1.8.4.3
And this happens only for polycom phones and not on cisco … any idea ?
I’m confused by your description of the problem and you haven’t provided any diagnostics.
Hello,
Let me put it all again,
When polycom phones are used it generated signalling problem but RTP is good through & fro, on cisco phone this is not the case, the signalling is proper.
When Phones are behind nat and asterisk server is on public ip and using the parameter nat=yes ,
when call is made from extension to extension and when pickedup the RTP is fine , suppose the caller disconnects the call the called party is not disconnected still, it disconnects after some time called party call is disconnected with 481 transaction does not exist.
What could be the reason behind that as it is only on polycom and not cisco.
core set debug 5
core set verbose 3
sip set debug on
Run the call, copy the console output (or configure to log full debugging output,
Minimally obscure authentication data and other sensitive information, and post the result. Asterisk provides a lot of debugging information, trying to answer something without it is difficult.