[Asterisk-Users] Call Leg/Transaction Does Not Exist

Ashling O'Driscoll ashling.odriscoll at cit.ie
Thu Nov 4 12:35:30 MST 2004


Hi,

Thanks for the reply. Yes I had left out the 's'(as I had copied from
the previous thread) but that is not the problem. I still have the
'call leg transaction does not exist' error. I have included the
debug sip messages below if that will help any bit. I read that this
error should have something got to do with a sip cancel message, an
incorrect invite message or the to header. Since I am not inviting
anyone and I dont cancel I dont think they apply. However I also
think my 'to' header syntax is ok....so any ideas?

Thanks again,
Aisling.

Sip read:
REGISTER sip:172.16.3.15 SIP/2.0
Via: SIP/2.0/UDP 172.16.3.13:12568
Max-Forwards: 70
From:
<sip:odriscolla at 172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla at 172.16.3.15>
Call-ID: 1933df16b5b546dca9374168f6f72c59 at 172.16.3.13
CSeq: 71 REGISTER
Contact: <sip:172.16.3.13:12568>;methods="INVITE, MESSAGE, INFO,
SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER"
User-Agent: RTC/1.2.4949 (Messenger 5.0.0482)
Event: registration
Allow-Events: presence
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 172.16.3.13 : 12568 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.3.13:12568
From:
<sip:odriscolla at 172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla at 172.16.3.15>;tag=as0442c120
Call-ID: 1933df16b5b546dca9374168f6f72c59 at 172.16.3.13
CSeq: 71 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:odriscolla at 172.16.3.15>
Content-Length: 0


 to 172.16.3.13:12568
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.3.13:12568
From:
<sip:odriscolla at 172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla at 172.16.3.15>;tag=as0442c120
Call-ID: 1933df16b5b546dca9374168f6f72c59 at 172.16.3.13
CSeq: 71 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:odriscolla at 172.16.3.15>
WWW-Authenticate: Digest realm="asterisk", nonce="09fbe581"
Content-Length: 0


 to 172.16.3.13:12568
Scheduling destruction of call
'1933df16b5b546dca9374168f6f72c59 at 172.16.3.13' in 15000 ms


Sip read:
REGISTER sip:172.16.3.15 SIP/2.0
Via: SIP/2.0/UDP 172.16.3.13:12568
Max-Forwards: 70
From:
<sip:odriscolla at 172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla at 172.16.3.15>
Call-ID: 1933df16b5b546dca9374168f6f72c59 at 172.16.3.13
CSeq: 72 REGISTER
Contact: <sip:172.16.3.13:12568>;methods="INVITE, MESSAGE, INFO,
SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER"
User-Agent: RTC/1.2.4949 (Messenger 5.0.0482)
Authorization: Digest username="odriscolla", realm="asterisk",
algorithm=md5, uri="sip:172.16.3.15", nonce="09fbe581",
response="488e7216327e85c4bc1976050ce81310"
Event: registration
Allow-Events: presence
Content-Length: 0


13 headers, 0 lines
Using latest request as basis request
Sending to 172.16.3.13 : 12568 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.3.13:12568
From:
<sip:odriscolla at 172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla at 172.16.3.15>;tag=as0442c120
Call-ID: 1933df16b5b546dca9374168f6f72c59 at 172.16.3.13
CSeq: 72 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:odriscolla at 172.16.3.15>
Content-Length: 0


 to 172.16.3.13:12568
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.3.13:12568
From:
<sip:odriscolla at 172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla at 172.16.3.15>;tag=as0442c120
Call-ID: 1933df16b5b546dca9374168f6f72c59 at 172.16.3.13
CSeq: 72 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:172.16.3.13:12568>;expires=120
Date: Thu, 04 Nov 2004 19:31:22 GMT
Content-Length: 0


 to 172.16.3.13:12568
Scheduling destruction of call
'1933df16b5b546dca9374168f6f72c59 at 172.16.3.13' in 15000 ms
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:172.16.3.13:12568 SIP/2.0
Via: SIP/2.0/UDP 172.16.3.15:5060;branch=z9hG4bK2284d025
From: "asterisk" <sip:asterisk at 172.16.3.15>;tag=as12bc656c
To: <sip:172.16.3.13:12568>
Contact: <sip:asterisk at 172.16.3.15>
Call-ID: 25ea0a776295312e72a4fcd845550d6b at 172.16.3.15
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 38

Messages-Waiting: no
Voicemail: 0/0
 (no NAT) to 172.16.3.13:12568
Scheduling destruction of call
'25ea0a776295312e72a4fcd845550d6b at 172.16.3.15' in 15000 ms


Sip read:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 172.16.3.15:5060;branch=z9hG4bK2284d025
From: "asterisk" <sip:asterisk at 172.16.3.15>;tag=as12bc656c
To: <sip:172.16.3.13:12568>;tag=a05ddee6260049778a66b59fb903130d
Call-ID: 25ea0a776295312e72a4fcd845550d6b at 172.16.3.15
CSeq: 102 NOTIFY
User-Agent: RTC/1.2
Content-Length: 0


8 headers, 0 lines
    -- Got SIP response 481 "Call Leg/Transaction Does Not Exist"
back from 172.16.3.13
Destroying call '25ea0a776295312e72a4fcd845550d6b at 172.16.3.15'
Destroying call '1933df16b5b546dca9374168f6f72c59 at 172.16.3.13'

---- Original Message ----
From: el_flynn at lanvik-icu.com
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Call Leg/Transaction Does Not Exist"
back
Date: Fri, 05 Nov 2004 03:16:13 +0800

>On 11/4/2004, "Ashling O'Driscoll" <ashling.odriscoll at cit.ie> wrote:
>
>>[general]
>>
>>port =3D 5060 ; Port to bind to (SIP is 5060)
>>bindaddr =3D 0=2E0=2E0=2E0 ; Address to bind to (all addresses on
>machine)=
>>
>>diallow=3Dall=20
>>allow=3Dulaw
>>context =3D from-sip ; Send SIP callers that we don't know about
>here
>>
>>;register =3D> 2000:suzuki at 172=2E16=2E3=2E15
>>
>
>the HTML posting sort of screwed up the content of your email, but if
>i
>interpret it correctly it looks like you've got a line that says
>
>diallow=all
>
>shouldn't that be
>
>disallow=all
>
>Flynn
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