Hello, I am having a problem with getting call transfer to work.
This is what is happening:- 1) External call comes in on SIP from a DDI provider 2) The call is answered by extension 204 3) Then extension 204 presses the Xfer button and the call is placed on hold 4) Extension 204 calls extension 201 and speaks to them. 5) Extension 204 presses the xfer button again to complete the transfer. The result is that the caller is cut off and the SIP Debug in asterisk shows the following:- SIP/2.0 481 Call leg/transaction does not exist Below is a clip from the debug list. I would greatly appreciate any help as the client is getting annoyed. Regards Dan <------------> -- Packet2Packet bridging SIP/winsor_204-12cb4160 and SIP/winsor_201-12ca50b0 sip1*CLI> <--- SIP read from 94.193.81.135:49160 ---> ACK sip:2...@83.222.226.126 SIP/2.0 Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-9ba5b149 From: "Rachael" <sip:winsor_...@sip1.keshercommunications.com>;tag=127e2c656448055eo0 To: "Robert" <sip:2...@sip1.keshercommunications.com>;tag=as1db0f5fd Call-ID: 5060f231-68791...@94.193.81.135 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="winsor_204",realm="asterisk",nonce="24eede11",uri="sip:2...@83. 222.226.126",algorithm=MD5,response="a3b443415fd656ce42253002548a823a" Contact: "Rachael" <sip:winsor_...@94.193.81.135:49160> User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 <-------------> --- (11 headers 0 lines) --- sip1*CLI> <--- SIP read from 94.193.81.135:49160 ---> REFER sip:901617720...@83.222.226.126 SIP/2.0 Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-5479aeea From: <sip:winsor_...@94.193.81.135:49160>;tag=f2c2287b333442fi0 To: "01617720007" <sip:901617720...@83.222.226.126>;tag=as2eb45d54 Referred-By: "Rachael" <sip:winsor_...@sip1.keshercommunications.com> Call-ID: 15dcfde333cdaf86302cb6490b04d...@83.222.226.126 CSeq: 102 REFER Max-Forwards: 70 Contact: "Rachael" <sip:winsor_...@94.193.81.135:49160> efer-To: <sip:2...@83.222.226.126?replaces=5060f231%2d68791a02%4010%2e0%2e0%2e204% 3Bfrom-tag%3D127e2c656448055eo0%3Bto-tag%3Das1db0f5fd> User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Call 15dcfde333cdaf86302cb6490b04d...@83.222.226.126 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 2...@winsor_phones by winsor_...@sip1.keshercommunications.com <--- Transmitting (NAT) to 94.193.81.135:49160 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-5479aeea;received=94.193.81.135 From: <sip:winsor_...@94.193.81.135:49160>;tag=f2c2287b333442fi0 To: "01617720007" <sip:901617720...@83.222.226.126>;tag=as2eb45d54 Call-ID: 15dcfde333cdaf86302cb6490b04d...@83.222.226.126 CSeq: 102 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:901617720...@83.222.226.126> Content-Length: 0 <------------> set_destination: Parsing <sip:winsor_...@94.193.81.135:49160> for address/port to send to set_destination: set destination to 94.193.81.135, port 49160 Reliably Transmitting (NAT) to 94.193.81.135:49160: NOTIFY sip:winsor_...@94.193.81.135:49160 SIP/2.0 Via: SIP/2.0/UDP 83.222.226.126:5060;branch=z9hG4bK2e10dade;rport From: "01617720007" <sip:901617720...@83.222.226.126>;tag=as2eb45d54 To: <sip:winsor_...@94.193.81.135:49160>;tag=f2c2287b333442fi0 Contact: <sip:901617720...@83.222.226.126> Call-ID: 15dcfde333cdaf86302cb6490b04d...@83.222.226.126 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "01617720007" <sip:901617720...@83.222.226.126>;privacy=off;screen=no 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, INFO Supported: replaces Content-Length: 49 SIP/2.0 481 Call leg/transaction does not exist
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