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

[Cary Fitch]

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


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