Sorry to bump this one... Anyone have any other ideas on it?
Regards Steven Davison Net Technial Solutions -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Davison Sent: 21 January 2010 08:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Xfer issue between DataCenter and User Site Thanks for the responses on this one.... David Gibbons: reinvite=no is set, as we need the asterisk box to maintain the audio for recording... (I believe even if we didn't have this option, MixMonitor would have the same effect anyway.) Peder: the firewall is integrated into the router, and is a Zyxel 660H-D1... which hasn't caused NAT issues in the past, but it is something that we can switch out and see if a different make/model has the same problem. In answer to your questions, the Data Center IP is the external address that has been 1 to 1 Nat'd to the internal address. The phone site has no static Nat in place for Sip or RTP, so we are reliant on the routers ability to sort that out. There is a firewall on that router, which allows ALL traffic out, and also allows SIP and RTP in. Hope that clears up a few things! :) Steven Davison - Network Engineer t: 0845 0034567 f: 0845 0034543 w: www.ntsols.com Net Technical Solutions | Suite 1 Wesley Chambers | Queens Road | Aldershot | Hampshire | GU11 3JD -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 20 January 2010 18:24 To: asterisk-users@lists.digium.com Cc: Alistair Mackenzie Subject: Re: [asterisk-users] Call Xfer issue between DataCenter and User Site Admittedly I didn't read your SIP debug (on the mobile), but do you have reinvite=no set for the extensions and SIP trunks (providers)? This sounds on the surface like a classic case of the Mondays. Erm reinvites I mean. <snip> 1. Incoming call from pstn/viop provider 2. Call is answered by a user 3. Call needs to be transferred 4. Xfer button is pushed, other user is called, answered, and they speak about the call 4b. The incoming call is held, listening to MoH 5. Xfer is pushed again, 6. <SIP Debug Output> 7. MoH stops, 8. Office user gets no audio 9. Incoming call is silent, and then call is dropped 10. Office user gets fed up of saying ‘hello??!?’ and hangs up. </snip> -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
