[asterisk-users] FW: Call Xfer issue between DataCenter and User Site

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

Re: [asterisk-users] Call Xfer issue between DataCenter and User Site

. 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

Re: [asterisk-users] DTMF reception during WaitForSilence

You last question : why are DTMF tones not audible in the recording? WE had issues with DTMF not recording, and found it was due to the handset only sending the DTMF in data, rather than inline, as a beep... that could be your reason :) Steven Davison Net Technial Solutions -Original

Re: [asterisk-users] Caller hang up not detected

Hi, Couple of questions... Are you allowing reinvites, and what happens if you change the dialplan to this? exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT) exten = 1,n,Playback(vm-goodbye) exten = 1,n,Hangup() help this helps :) Steven Davison Net Technial Solutions From: asterisk

[asterisk-users] Call Xfer issue between DataCenter and User Site

to anyone who may have ideas for this... ☺ 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