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:201 at 83.222.226.126 SIP/2.0
Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-9ba5b149
From: "Rachael"
<sip:winsor_204 at sip1.keshercommunications.com>;tag=127e2c656448055eo0
To: "Robert" <sip:201 at
sip1.keshercommunications.com>;tag=as1db0f5fd
Call-ID: 5060f231-68791a02 at 94.193.81.135
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest
username="winsor_204",realm="asterisk",nonce="24eede11",uri="sip:201
at 83.
222.226.126",algorithm=MD5,response="a3b443415fd656ce42253002548a823a"
Contact: "Rachael" <sip:winsor_204 at 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:901617720007 at 83.222.226.126 SIP/2.0
Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-5479aeea
From: <sip:winsor_204 at 94.193.81.135:49160>;tag=f2c2287b333442fi0
To: "01617720007" <sip:901617720007 at
83.222.226.126>;tag=as2eb45d54
Referred-By: "Rachael" <sip:winsor_204 at
sip1.keshercommunications.com>
Call-ID: 15dcfde333cdaf86302cb6490b04d745 at 83.222.226.126
CSeq: 102 REFER
Max-Forwards: 70
Contact: "Rachael" <sip:winsor_204 at 94.193.81.135:49160>
efer-To:
<sip:201 at 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 15dcfde333cdaf86302cb6490b04d745 at 83.222.226.126 got a SIP call
transfer from caller: (REFER)!
SIP transfer to extension 201 at winsor_phones by
winsor_204 at 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_204 at 94.193.81.135:49160>;tag=f2c2287b333442fi0
To: "01617720007" <sip:901617720007 at
83.222.226.126>;tag=as2eb45d54
Call-ID: 15dcfde333cdaf86302cb6490b04d745 at 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:901617720007 at 83.222.226.126>
Content-Length: 0
<------------>
set_destination: Parsing <sip:winsor_204 at 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_204 at 94.193.81.135:49160 SIP/2.0
Via: SIP/2.0/UDP 83.222.226.126:5060;branch=z9hG4bK2e10dade;rport
From: "01617720007" <sip:901617720007 at
83.222.226.126>;tag=as2eb45d54
To: <sip:winsor_204 at 94.193.81.135:49160>;tag=f2c2287b333442fi0
Contact: <sip:901617720007 at 83.222.226.126>
Call-ID: 15dcfde333cdaf86302cb6490b04d745 at 83.222.226.126
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "01617720007"
<sip:901617720007 at 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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