[Asterisk-Users] Call Leg/Transaction Does Not Exist
Richard
richard at o-matrix.org
Thu Nov 4 12:45:15 MST 2004
You get the 481 error because * sends a NOTIFY message for MWI. The client,
Microsoft messenger doesn't support it and sends the error back.
Sip NOTIFY has several usages. MWI is one of them. MS-messenger probably
supports other usages of NOTIFY but not for MWI.
Richard
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Ashling O'Driscoll
> Sent: Thursday, November 04, 2004 9:36 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Call Leg/Transaction Does Not Exist
>
>
> 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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