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: [email protected] > [mailto:[email protected]] On Behalf Of Yves Arikoglu > Sent: Monday, January 25, 2010 7:28 AM > To: [email protected] > 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 > > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
