Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem
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-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu Sent: Monday, January 25, 2010 7:28 AM To: asterisk-users@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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem
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-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu Sent: Monday, January 25, 2010 7:28 AM To: asterisk-users@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 -- _ -- 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
