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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-9ba5b149
From: "Rachael"
<sip:[email protected]>;tag=127e2c656448055eo0
To: "Robert" <sip:[email protected]>;tag=as1db0f5fd
Call-ID: [email protected]
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:[email protected]: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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 94.193.81.135:49160;branch=z9hG4bK-5479aeea
From: <sip:[email protected]:49160>;tag=f2c2287b333442fi0
To: "01617720007" <sip:[email protected]>;tag=as2eb45d54
Referred-By: "Rachael" <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 REFER
Max-Forwards: 70
Contact: "Rachael" <sip:[email protected]:49160>
efer-To:
<sip:[email protected]?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 [email protected] got a SIP call
transfer from caller: (REFER)!
SIP transfer to extension 2...@winsor_phones by
[email protected]
<--- 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:[email protected]:49160>;tag=f2c2287b333442fi0
To: "01617720007" <sip:[email protected]>;tag=as2eb45d54
Call-ID: [email protected]
CSeq: 102 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
set_destination: Parsing <sip:[email protected]: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:[email protected]:49160 SIP/2.0
Via: SIP/2.0/UDP 83.222.226.126:5060;branch=z9hG4bK2e10dade;rport
From: "01617720007" <sip:[email protected]>;tag=as2eb45d54
To: <sip:[email protected]:49160>;tag=f2c2287b333442fi0
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "01617720007"
<sip:[email protected]>;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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