[asterisk-users] asterisk 1.8 sends "SIP/2.0 481 Call/Transaction Does Not Exist" to INVITE

[Miguel Oyarzo]

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."
>
>
> --
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