I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc.
However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is sip debug from pressing "transfer->blind->dial dest->Dial Key" (note both stations do have access tot eh dial-dst ext of 202010) <------------> -- Started music on hold, class 'default', on channel 'SIP/1050-0a6ffa70' <--- SIP read from XXX.XXX.232.66:8986 ---> ACK sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bK3dc3ce44DF7EE83D From: "1051" <sip:1...@xxx.xxx.232.66:8986>;tag=D117C080-6FFBC539 To: "1050" <sip:1...@xxx.xxx.232.175>;tag=as140f4415 CSeq: 1 ACK Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: <sip:1...@xxx.xxx.232.66:8986> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from XXX.XXX.232.66:8986 ---> REFER sip:1...@xxx.xxx.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A From: "1051" <sip:1...@xxx.xxx.232.66:8986>;tag=D117C080-6FFBC539 To: "1050" <sip:1...@xxx.xxx.232.175>;tag=as140f4415 CSeq: 2 REFER Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 Contact: <sip:1...@xxx.xxx.232.66:8986> User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Refer-To: sip:202...@xxx.xxx.232.175;user=phone Referred-By: <sip:1...@xxx.xxx.232.175> Max-Forwards: 70 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Call 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 got a SIP call transfer from caller: (REFER)! <--- Transmitting (no NAT) to XXX.XXX.232.66:8986 ---> SIP/2.0 603 Declined (policy) Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A;received=XXX.XXX.232.66 From: "1051" <sip:1...@xxx.xxx.232.66:8986>;tag=D117C080-6FFBC539 To: "1050" <sip:1...@xxx.xxx.232.175>;tag=as140f4415 Call-ID: 75235b51626f63440c264f6b70dc5...@xxx.xxx.232.175 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:1...@xxx.xxx.232.175> Content-Length: 0 <------------> -- Stopped music on hold on SIP/1050-0a6ffa70 _______________________________________________ -- 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