Ashling O'Driscoll
2004-Nov-04 12:35 UTC
[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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla@172.16.3.15>
Call-ID: 1933df16b5b546dca9374168f6f72c59@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla@172.16.3.15>;tag=as0442c120
Call-ID: 1933df16b5b546dca9374168f6f72c59@172.16.3.13
CSeq: 71 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:odriscolla@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla@172.16.3.15>;tag=as0442c120
Call-ID: 1933df16b5b546dca9374168f6f72c59@172.16.3.13
CSeq: 71 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:odriscolla@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@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla@172.16.3.15>
Call-ID: 1933df16b5b546dca9374168f6f72c59@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla@172.16.3.15>;tag=as0442c120
Call-ID: 1933df16b5b546dca9374168f6f72c59@172.16.3.13
CSeq: 72 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:odriscolla@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid
=f9d3957208
To: <sip:odriscolla@172.16.3.15>;tag=as0442c120
Call-ID: 1933df16b5b546dca9374168f6f72c59@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@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@172.16.3.15>;tag=as12bc656c
To: <sip:172.16.3.13:12568>
Contact: <sip:asterisk@172.16.3.15>
Call-ID: 25ea0a776295312e72a4fcd845550d6b@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@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@172.16.3.15>;tag=as12bc656c
To: <sip:172.16.3.13:12568>;tag=a05ddee6260049778a66b59fb903130d
Call-ID: 25ea0a776295312e72a4fcd845550d6b@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@172.16.3.15'
Destroying call '1933df16b5b546dca9374168f6f72c59@172.16.3.13'
---- Original Message ----
From: el_flynn@lanvik-icu.com
To: asterisk-users@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@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@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
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>-------------------Legal
>Disclaimer---------------------------------------
>
>The above electronic mail transmission is confidential and intended
>only for the person to whom it is addressed. Its contents may be
>protected by legal and/or professional privilege. Should it be
>received by you in error please contact the sender at the above
>quoted email address. Any unauthorised form of reproduction of this
>message is strictly prohibited. The Institute does not guarantee the
>security of any information electronically transmitted and is not
>liable if the information contained in this communication is not a
>proper and complete record of the message as transmitted by the
>sender nor for any delay in its receipt.
>
>---------------------------------------------------------------------
>-------------------
-------------------Legal Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only for the
person to whom it is addressed. Its contents may be protected by legal and/or
professional privilege. Should it be received by you in error please contact the
sender at the above quoted email address. Any unauthorised form of reproduction
of this message is strictly prohibited. The Institute does not guarantee the
security of any information electronically transmitted and is not liable if the
information contained in this communication is not a proper and complete record
of the message as transmitted by the sender nor for any delay in its receipt.
----------------------------------------------------------------------------------------
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@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Ashling O'Driscoll > Sent: Thursday, November 04, 2004 9:36 AM > To: asterisk-users@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid > =f9d3957208 > To: <sip:odriscolla@172.16.3.15> > Call-ID: 1933df16b5b546dca9374168f6f72c59@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid > =f9d3957208 > To: <sip:odriscolla@172.16.3.15>;tag=as0442c120 > Call-ID: 1933df16b5b546dca9374168f6f72c59@172.16.3.13 > CSeq: 71 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:odriscolla@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid > =f9d3957208 > To: <sip:odriscolla@172.16.3.15>;tag=as0442c120 > Call-ID: 1933df16b5b546dca9374168f6f72c59@172.16.3.13 > CSeq: 71 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:odriscolla@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@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid > =f9d3957208 > To: <sip:odriscolla@172.16.3.15> > Call-ID: 1933df16b5b546dca9374168f6f72c59@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid > =f9d3957208 > To: <sip:odriscolla@172.16.3.15>;tag=as0442c120 > Call-ID: 1933df16b5b546dca9374168f6f72c59@172.16.3.13 > CSeq: 72 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:odriscolla@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@172.16.3.15>;tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid > =f9d3957208 > To: <sip:odriscolla@172.16.3.15>;tag=as0442c120 > Call-ID: 1933df16b5b546dca9374168f6f72c59@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@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@172.16.3.15>;tag=as12bc656c > To: <sip:172.16.3.13:12568> > Contact: <sip:asterisk@172.16.3.15> > Call-ID: 25ea0a776295312e72a4fcd845550d6b@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@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@172.16.3.15>;tag=as12bc656c > To: <sip:172.16.3.13:12568>;tag=a05ddee6260049778a66b59fb903130d > Call-ID: 25ea0a776295312e72a4fcd845550d6b@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@172.16.3.15' > Destroying call '1933df16b5b546dca9374168f6f72c59@172.16.3.13' > > ---- Original Message ---- > From: el_flynn@lanvik-icu.com > To: asterisk-users@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@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@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 > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >-------------------Legal > >Disclaimer--------------------------------------- > > > >The above electronic mail transmission is confidential and intended > >only for the person to whom it is addressed. Its contents may be > >protected by legal and/or professional privilege. Should it be > >received by you in error please contact the sender at the above > >quoted email address. Any unauthorised form of reproduction of this > >message is strictly prohibited. The Institute does not guarantee the > >security of any information electronically transmitted and is not > >liable if the information contained in this communication is not a > >proper and complete record of the message as transmitted by the > >sender nor for any delay in its receipt. > > > >--------------------------------------------------------------------- > >------------------- > > > -------------------Legal Disclaimer-------------------------------------- > - > > The above electronic mail transmission is confidential and intended only > for the person to whom it is addressed. Its contents may be protected by > legal and/or professional privilege. Should it be received by you in error > please contact the sender at the above quoted email address. Any > unauthorised form of reproduction of this message is strictly prohibited. > The Institute does not guarantee the security of any information > electronically transmitted and is not liable if the information contained > in this communication is not a proper and complete record of the message > as transmitted by the sender nor for any delay in its receipt. > > -------------------------------------------------------------------------- > -------------- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users