[asterisk-users] sip.conf with versatel and two NICs very strangeproblem

[Yves Arikoglu]

thanks, i tried this already.... but unfortunately no change.
any further suggestions or answers concerning my other questions?

thanx, yves

Cary Fitch schrieb:
> As a guess, they can both talk to the server, but can't talk to each other.
>
>
> What is common to that is they may be trying to reinvite each other, and
> there is no path through the respective routers/firewalls to the other.
>
> So if reinvite is set to yes, set it to no, in both phone profiles on the
> server.
>
> Cary Fitch
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Yves Arikoglu
> Sent: Monday, January 25, 2010 7:28 AM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] sip.conf with versatel and two NICs very
> strangeproblem
>
> Hi
>
> My System is:
> Asterisk 1.6 running on a Dell Server with two network interfaces.
> eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has 
> the local ip 10.26.208.252
> and the external ip 89.244.x.y
>
> eth0 of the server is configured to 10.26.192.107
>
> The Problem:
> SIP registration works, phone rings in- and outbound, but there is no 
> audio, nor the caller neither the callee
> can hear anything.
> So i am quite sure that is has something to do with firewalls, natting 
> and so on but i?ve read hundreds of
> pages and tried thousands of setting but i cant get audio to work..
> the strange thing is... when i call the versatel-sip-number from my 
> mobile phone, i see the call coming in
> in the cli, i see the voiceprompts that asterisk plays, but even there I 
> cant hear anything on my mobile.
> next strange thing:
> i defined 2 sip-extensions. both are registered... everything is fine... 
> routes are ok, they can call out
> and can be called from external and from internal (sip phones call each 
> other).. but the same... no audio.
> but when one sip extension calls a wrong number... the "cannot be 
> completed" message is hearable.
> i configured a queue with moh and even this works... but why cant to 
> sip-phones talk to each other?
> why cant an external caller hear any audio?
>
> if i make sip debug, i see traffic (and due to extension is calling i 
> think that on the sip-level everything
> is okay...) how can i see, which port and interface is chosen for audio 
> when a call comes in?
>
> thanks,
> yves
>
>
>