[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 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
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
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 Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu Sent: 21 January 2010 11:00 To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF reception during WaitForSilence Hello, I wrote a little AGI-Script that implements an IVR (using asterisk 1.6). The whole conversation is recorded and at some points the caller should tell some information. I detect the silence (WaitForSilence) to go to the next step in the IVR. Until now everything is OK, but... some information the user gives (or speaks) is numeric... some users have the habit, to enter numeric information via the phonekeypad (ergo creating dtmf-tones) but I cant process DTMF-Input during WaitForSilence. How can I achive that both works simultaneously? I mean recording the spoken digits AND detecting DTMF-Input AND detecting silence to know, when Input has finished... (I want to avoid that users have to finish their input with the pound-key...) ? Btw.: why are the DTMF-Tones, that a user enters, not hearable in the recording? Thanks for your help and hints, 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] 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-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hugolivude Sent: 21 January 2010 13:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller hang up not detected Hi, I'm having trouble getting Dial to exit when the caller hangs up in Asterisk 1.4.21.2. I use a POTS line to call into the DiD given to me by VOIP service provider. When the call comes in, I have the VOIP provider send it to another POTS line. All this works fine however when the caller (me) hangs up, the Dial command does not exit. The callee stays connected (and my billing continues!). Dial doesn't exit until the callee hangs up. Here's a snip from extensions.conf: exten = 1,n,Dial(SIP/14168724...@6135551212-sw1|120|gtT) exten = 1,n,Playback(vm-goodbye) Here's the CLI output (verbosity = 4): -- Executing [...@trunk-0001:1] NoOp(SIP/77.57.127.163-09023590, ) in new stack -- Executing [...@trunk-0001:2] Dial(SIP/77.57.127.163-09023590, SIP/14168724...@6135551212-sw1|120|gtT) in new stack -- Called 14168724...@6135551212-sw1 -- SIP/6135551212-sw1-090275d0 is making progress passing it to SIP/77.57.127.163-09023590 -- SIP/6135551212-sw1-090275d0 answered SIP/77.57.127.163-09023590 *** I hang up here, but the call continues. A while later the callee hangs up: -- Executing [...@trunk-0001:3] Playback(SIP/77.57.127.163-09023590, vm-goodbye) in new stack *** obviously I don't here this, just see it in the CLI I'd be grateful for any troubleshooting tips that will help me get asterisk to quit the Dial command when the originator hangs up. Thanks, H -- _ -- 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
[asterisk-users] Call Xfer issue between DataCenter and User Site
Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat’d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this correctly. The handsets are Linksys SPA922 The issue we are getting is in transferring calls, which happens like this :- 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. Here is the sip debug output... [Jan 20 16:43:38] set_destination: Parsing sip:1...@xxx.xxx.xxx.xxx:10036 for address/port to send to [Jan 20 16:43:38] set_destination: set destination to XXX.XXX.XXX.XXX, port 10036 [Jan 20 16:43:38] Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:10016: NOTIFY sip:1...@xxx.xxx.xxx.xxx:10036 SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK48cff632;rport Max-Forwards: 70 From: Steve (NetTech) sip:1...@yyy.yyy.yyy.yyy;tag=as4f7c4d0c To: sip:1...@xxx.xxx.xxx.xxx:10036;tag=726be2fb618280d0i0 Contact: sip:1...@yyy.yyy.yyy.yyy Call-ID: 718a30a4572984a918b88dc64df64...@yyy.yyy.yyy.yyy CSeq: 103 NOTIFY User-Agent: Asterisk PBX 1.6.1.1 Event: refer;id=102 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 49 SIP/2.0 481 Call leg/transaction does not exist --- [Jan 20 16:43:38] -- Stopped music on hold on SIP/176-09bf9630 [Jan 20 16:43:38] --- SIP read from UDP://XXX.XXX.XXX.XXX:10016 --- SIP/2.0 200 OK To: sip:1...@xxx.xxx.xxx.xxx:10036;tag=726be2fb618280d0i0 From: Steve (NetTech) sip:1...@yyy.yyy.yyy.yyy;tag=as4f7c4d0c Call-ID: 718a30a4572984a918b88dc64df64...@yyy.yyy.yyy.yyy CSeq: 103 NOTIFY Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK48cff632 Server: Linksys/SPA922-4.1.18 Content-Length: 0 - YYY.YYY.YYY.YYY is the IP of the Datacenter XXX.XXX.XXX.XXX is the IP of the Office I have been going over and over the configs on the routers, sip.conf etc trying to work this out... we have also checked that the users are using the above sequence to transfer a call... Thanks 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 -- _ -- 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
