Vik Killa
2013-Sep-16 20:18 UTC
[asterisk-users] asterisk 1.8 sends "SIP/2.0 481 Call/Transaction Does Not Exist" to INVITE
Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> To: <sip:8009499014 at X.YYY.32.10 :5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898 Via: SIP/2.0/UDP X.YYY.32.10;branch=z9hG4bK542e.5042d534.0 Via: SIP/2.0/UDP X.YYY.33.178:5060;rport=5060;received=X.YYY.33.178;branch=z9hG4bK57b720cccb00f8498662f48e8 Call-ID: 94f80f866e877490729548a079abe371 at 192.168.101.5 CSeq: 2 INVITE Contact: <sip:7166359474 at X.YYY.33.178:5060> Max-Forwards: 69 x-inin-crn: 2001471530;loc=Amherst;ms=STAMPEDE-MS Supported: join, replaces User-Agent: ININ-TsServer/3.13.11.12748 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, SUBSCRIBE Accept: application/sdp Accept-Encoding: identity Content-Type: application/sdp Content-Length: 252 Proxy-Authorization: Digest username="909003660716",realm="X.YYY.32.10",nonce="5237559000011a22ed0fae66765d46ef9131e311fbb9d2fb",uri="sip:8009499014 at X.YYY.32.10 :5060",response="cb6306569b3047ac35064dcb5aee6db4" X-Enswitch-RURI: sip:8009499014 at X.YYY.32.10:5060 X-Enswitch-Source: X.YYY.33.178:5060 The only problem I see with this INVITE is the VIAs are not right after the INVITE line... although in https://www.ietf.org/rfc/rfc3261.txt, it explicitly states the the order of the headers is not a requirement, it seems Asterisk does make it one... "The relative order of header fields with different field names is not significant. However, it is RECOMMENDED that header fields which are needed for proxy processing (Via, Route, Record-Route, Proxy-Require, Max-Forwards, and Proxy-Authorization, for example) appear towards the top of the message to facilitate rapid parsing. The relative order of header field rows with the same field name is important." -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130916/ad83390f/attachment.htm>
Miguel Oyarzo
2013-Sep-16 21:31 UTC
[asterisk-users] asterisk 1.8 sends "SIP/2.0 481 Call/Transaction Does Not Exist" to INVITE
To: <sip:8009499014 at X.YYY.32.10:5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 In your call sample To has a tag. if this is the first Invite it can't have a tag at the end, otherwise Asterisk will look for an existing dialog in its database and will show an error, if can't find any. It looks like the other end is never closing the previous dialog?.. is Asterisk sending a proper 200 OK after receiving a BYE? NAT problem? regards, -- =================================Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 9/17/2013 6:18 AM, Vik Killa wrote:> Asterisk is sending a 481 in response to an INVITE for reasons I do > not understand. Here is the INVITE: > > > INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 > Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> > To: > <sip:8009499014 at X.YYY.32.10:5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 > From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898 > Via: SIP/2.0/UDP X.YYY.32.10;branch=z9hG4bK542e.5042d534.0 > Via: SIP/2.0/UDP > X.YYY.33.178:5060;rport=5060;received=X.YYY.33.178;branch=z9hG4bK57b720cccb00f8498662f48e8 > Call-ID: 94f80f866e877490729548a079abe371 at 192.168.101.5 > <mailto:94f80f866e877490729548a079abe371 at 192.168.101.5> > CSeq: 2 INVITE > Contact: <sip:7166359474 at X.YYY.33.178:5060> > Max-Forwards: 69 > x-inin-crn: 2001471530;loc=Amherst;ms=STAMPEDE-MS > Supported: join, replaces > User-Agent: ININ-TsServer/3.13.11.12748 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, > SUBSCRIBE > Accept: application/sdp > Accept-Encoding: identity > Content-Type: application/sdp > Content-Length: 252 > Proxy-Authorization: Digest > username="909003660716",realm="X.YYY.32.10",nonce="5237559000011a22ed0fae66765d46ef9131e311fbb9d2fb",uri="sip:8009499014 at X.YYY.32.10:5060",response="cb6306569b3047ac35064dcb5aee6db4" > X-Enswitch-RURI: sip:8009499014 at X.YYY.32.10:5060 > X-Enswitch-Source: X.YYY.33.178:5060 > > > > The only problem I see with this INVITE is the VIAs are not right > after the INVITE line... although in > https://www.ietf.org/rfc/rfc3261.txt, it explicitly states the the > order of the headers is not a requirement, it seems Asterisk does make > it one... > > "The relative order of header fields with different field names is not > significant. However, it is RECOMMENDED that header fields which are > needed for proxy processing (Via, Route, Record-Route, Proxy-Require, > Max-Forwards, and Proxy-Authorization, for example) appear towards > the top of the message to facilitate rapid parsing. The relative > order of header field rows with the same field name is important." > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130917/0a8c7cae/attachment.htm>