[OpenSIPS-Users] BYE - 404 not here

Max Mühlbronner mm at 42com.com
Thu Feb 4 16:11:01 CET 2010


sorry, of course, here are the invites:


*Invite from asterisk --> opensips*

U 62.66.66.67:5060 -> 62.66.66.66:5060
INVITE sip:1235493333316 at 62.66.66.66 SIP/2.0.
Via: SIP/2.0/UDP 62.66.66.67:5060;branch=z9hG4bK16ff74c2;rport.
Max-Forwards: 70.
From: "49302332434343" <sip:49302332434343 at 62.66.66.67>;tag=as1fcd8c32.
To: <sip:1235493333316 at 62.66.66.66>.
Contact: <sip:49302332434343 at 62.66.66.67>.
Call-ID: 7061c14c469285fc782f31d128791bc5 at 62.66.66.67.
CSeq: 102 INVITE.
User-Agent: INES.
Date: Thu, 04 Feb 2010 10:33:10 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 290.
.
v=0.
o=root 992341641 992341641 IN IP4 62.66.66.67.
s=Asterisk PBX 1.6.0-beta9.
c=IN IP4 62.66.66.67.
t=0 0.
m=audio 32482 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.



*Invite from opensips --> pstngw:

*
U 62.66.66.66:5060 ->  213.20.11.11:5060
INVITE sip:493333316 at 213.20.11.11 SIP/2.0.
Record-Route: <sip:62.66.66.66;lr=on;ftag=as1fcd8c32;did=a1.547e8c16>.
Via: SIP/2.0/UDP 62.66.66.66;branch=z9hG4bKbee2.9124c5b3.0.
Via: SIP/2.0/UDP 
62.66.66.67:5060;received=62.66.66.67;branch=z9hG4bK16ff74c2;rport=5060.
Max-Forwards: 69.
From: "49302332434343" <sip:49302332434343 at 62.66.66.67>;tag=as1fcd8c32.
To: <sip:1235493333316 at 62.66.66.66>.
Contact: <sip:49302332434343 at 62.66.66.67>.
Call-ID: 7061c14c469285fc782f31d128791bc5 at 62.66.66.67.
CSeq: 102 INVITE.
User-Agent: INES.
Date: Thu, 04 Feb 2010 10:33:10 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 308.
Session-Expires: 1800.
.
v=0.
o=root 992341641 992341641 IN IP4 62.66.66.66.
s=Asterisk PBX 1.6.0-beta9.
c=IN IP4 62.66.66.66.
t=0 0.
m=audio 55408 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
a=nortpproxy:yes.


Hope you can find something useful. Thanks.


<mailto:sip%3A1235493333316 at 62.66.66.66>Brett Nemeroff schrieb:
> >From my experience, this usually happen either from a configuration 
> file error, or from the terminating UAS who sends the  BYE not sending 
> it to the RURI in the contact header from the original INVITE. Can we 
> see the original INVITE as it hits OpenSIPs?
> -Brett
>
>
> On Thu, Feb 4, 2010 at 4:56 AM, Max Mühlbronner <mm at 42com.com 
> <mailto:mm at 42com.com>> wrote:
>
>     Hello everyone,
>
>     i have a problem when a call is hangup by the callee, i think i
>     probably
>     have some general routing logic Problem and i cant find any way to
>     solve it.
>
>     caller --> asterisk (62.66.66.67) --> opensips(62.66.66.66) (+rtpproxy
>     on the same machine) --> pstngw (213.20.11.11)
>
>     Everything seems to be working fine, i have been testing a long time
>     but  i recognized some problem. When the callee rejects the call ,
>     (486
>     busy) the busy is fine, transmitted back to the caller.
>     But if the call is established and the callee hangs up, the BYE is not
>     received by the original calling side so it stays connected.
>     My opensips knowledge is still very basic, so please excuse if it is
>     some dumb routing mistake made by me.
>
>     62.66.66.66 --> opensips
>     62.66.66.67 --> asterisk
>     213.20.11.11 --> pstngw
>
>
>     The busy is fine, and transmitted correctly (and callattempt is
>     stopped), but the bye is not received on the side where the call was
>     originating from (asterisk).
>
>
>     U 213.20.11.11:5060 <http://213.20.11.11:5060> -> 62.66.66.66:5060
>     <http://62.66.66.66:5060>
>     SIP/2.0 486 Busy Here.
>     Via: SIP/2.0/UDP 62.66.66.66;branch=z9hG4bKf41a.b1b6523.0.
>     Via: SIP/2.0/UDP
>     62.66.66.67:5060;received=62.66.66.67;branch=z9hG4bK79996cc9;rport=5060.
>     From: "49302332434343" <sip:49302332434343 at 62.66.66.67
>     <mailto:sip%3A49302332434343 at 62.66.66.67>>;tag=as67e89fcd.
>     To: <sip:1235493333316 at 62.66.66.66
>     <mailto:sip%3A1235493333316 at 62.66.66.66>>;tag=255533104.
>     Call-ID: 2da8e3f7156077231d448a3530a87a21 at 62.66.66.67
>     <mailto:2da8e3f7156077231d448a3530a87a21 at 62.66.66.67>.
>     CSeq: 102 INVITE.
>     Contact: <sip:493333316 at whs1.bla.voip.bels.com:5060
>     <http://sip:493333316@whs1.bla.voip.bels.com:5060>>.
>     Content-Length: 0.
>
>
>     ---------------------
>
>
>     the not working BYE, followed by 404 not here, which is sent by the
>     basic routing block (like in most of the example scripts /
>     sl_send_reply("404","Not here");)
>
>
>     U 213.20.11.11:5060 <http://213.20.11.11:5060> -> 62.66.66.66:5060
>     <http://62.66.66.66:5060>
>     BYE sip:49302332434343 at 62.66.66.67
>     <mailto:sip%3A49302332434343 at 62.66.66.67> SIP/2.0.
>     Via: SIP/2.0/UDP
>     213.20.11.11:5060;branch=z9hG4bK623vjs00c0q1bggou101sd0000g00
>     .1.
>     From: <sip:1235493333316 at 213.20.11.11
>     <mailto:sip%3A1235493333316 at 213.20.11.11>>;tag=960392687.
>     To: "49302332434343" <sip:49302332434343 at 62.66.66.66
>     <mailto:sip%3A49302332434343 at 62.66.66.66>>;tag=as32838038.
>     Call-ID: 150c32ca5a8adcb8589445f95e9ffe34 at 62.66.66.67
>     <mailto:150c32ca5a8adcb8589445f95e9ffe34 at 62.66.66.67>.
>     CSeq: 1 BYE.
>     Max-Forwards: 9.
>     Supported: timer.
>     Content-Length: 0.
>     Route: <sip:62.66.66.66;lr=on;ftag=as32838038;did=fee.ee87c634>.
>
>
>     U 62.66.66.66:5060 <http://62.66.66.66:5060> -> 213.20.11.11:5060
>     <http://213.20.11.11:5060>
>     SIP/2.0 404 Not here.
>     Via: SIP/2.0/UDP
>     213.20.11.11:5060;rport=5060;received=213.20.11.11;branch=z9hG4bKk4ksmf10eosgjf41m3k0sd0000g00.1.
>     From: <sip:1235493333316 at 213.20.11.11
>     <mailto:sip%3A1235493333316 at 213.20.11.11>>;tag=1126364538.
>     To: "49302332434343" <sip:49302332434343 at 62.66.66.66
>     <mailto:sip%3A49302332434343 at 62.66.66.66>>;tag=as1fcd8c32.
>     Call-ID: 7061c14c469285fc782f31d128791bc5 at 62.66.66.67
>     <mailto:7061c14c469285fc782f31d128791bc5 at 62.66.66.67>.
>     CSeq: 1 BYE.
>     Server: OpenSIPS (1.6.1-notls (i386/linux)).
>     Content-Length: 0.
>
>
>
>
>     Thanks very much for any help, really appreciated. :)
>
>
>     Best Regards
>
>     Max M.
>
>
>     _______________________________________________
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>
>
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