Le 02/06/2012 19:18, Administrator TOOTAI a écrit :
Le 30/05/2012 15:02, Andres a écrit :
Considering that you made progress on your initial problem by setting
"nat=force_rport" (resulting in connected calls with no audio) and now
you're mentioning the use of "externaddr", I'd recommend a very
careful reading of the "NAT SUPPORT" section of sip.conf.sample in the
configs directory of the Asterisk source tree. In Asterisk 1.8, there
is a new configuration option named "media_address" which may be of
particular interest.
It sounds like a NAT issue to me too. Why don't you do a quick test
and put the Asterisk box on a public IP if you can. If it works, you
will have narrowed down the issue to a NAT problem. You could have
a nat router with a broken SIP ALG.
Back to the story: even out of VM -which means on a public IP- the
timeout problem till appears. And more odd, if a communication start,
the call get hanged up because of this timeout :-(
All peers and users are setted with nat=yes, phones connected to
Asterisk have directmedia=nonat and peers gateways have directmedia=yes.
Remember, we only face this problem with Dellmont services and
asterisk 1.8/10. Previous asterisk versions are working well.
Does someone else use Dellmont services (VoipBuster, SipDiscount,
Intenetcalls, Voicetrading, ...) with asterisk 1.8 or 10? If yes and
without problem, would it be possible to share configurations?
Thanks for your help.
For the archives.
Problem was with Dellmont services: no audio or calls stopping after 120
seconds. They gave me another IP for setting outgoing calls and now
everything is going smoothly with both versions.
Thanks for help.
--
Daniel
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