[asterisk-users] Call Transfer Problem

[Dan Journo]

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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