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11/04/2009 at 10:26AM PST, ID: 24871878
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8.2

Call leg/transaction does not exist

Asked by DanJourno in Asterisk Open Source Telephony, Voice Over IP, IP PBX Systems

Tags: asterisk, call transfer

Hi,

I'm having a problem with Asterisk and the extensions that are behind a router (NAT).

I have attached a snip of the logs from an asterisk debug. (If you need more, please let me know)

The log shows, when a call comes in from a DDI, an extension (ext. 204) within the office answers the call. (The asterisk server is out of the office in a data centre). The call is then put on hold by using the Xfer button on the phone and then the operator dials another extention (ext 201) which is within the same office and therefore behind the same NAT, they are able to speak to the other member of staff, however, when they press Xfer to complete the attended transfer, all the calls get disconnected.
The SIP debug in Asterisk shows:-

SIP/2.0 481 Call leg/transaction does not exist


It is really important that I resolve this issue as my manager is getting annoyed.

Any help would be greatly appreciated.
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<--- 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@94.193.81.135:49160>;tag=f2c2287b333442fi0
To: "01617720007" <sip:901617720007@83.222.226.126>;tag=as2eb45d54
Call-ID: 15dcfde333cdaf86302cb6490b04d745@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@83.222.226.126>
Content-Length: 0 

<------------>
set_destination: Parsing <sip:winsor_204@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@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@83.222.226.126>;tag=as2eb45d54
To: <sip:winsor_204@94.193.81.135:49160>;tag=f2c2287b333442fi0
Contact: <sip:901617720007@83.222.226.126>
Call-ID: 15dcfde333cdaf86302cb6490b04d745@83.222.226.126
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "01617720007" <sip:901617720007@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
[+][-]11/04/09 08:28 PM, ID: 25746776

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[+][-]11/05/09 03:17 AM, ID: 25748343

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[+][-]11/05/09 05:42 AM, ID: 25749289

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About this solution

Zones: Asterisk Open Source Telephony, Voice Over IP, IP PBX Systems
Tags: asterisk, call transfer
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Solution Provided By: DanJourno
Participating Experts: 2
Solution Grade: A
 
 
 
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20091111-EE-VQP-91 - Hierarchy / EE_QW_3_20080625