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
