Le 30/05/2012 15:02, Andres a écrit :
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.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.
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.
--
Daniel
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