[asterisk-users] Asterisk replying 491
Stefan Schmidt
sst at sil.at
Thu Oct 20 08:03:38 CDT 2011
Hello,
if this is the complete sip trace the UAS (client) have to reply with an
ACK after the 401 response from asterisk. thats why asterisk thinks the
request is still alive.
regards
stefan
Am 19.10.11 22:22, schrieb markus_weiler at mailworks.org:
> Hallo,
>
> any idea what's wrong with that invite??
> help would be greatly appreciated!
>
> thanks
>
> Markus
>
>
> U XX.199.123.185:5060 -> XX.189.169.66:5060
> INVITE sip:07111234567 at XX.189.169.66 SIP/2.0..Via: SIP/2.0/UDP
> 192.168.178.26:5060;rport;branch=z9hG4bK98099..Max-Forwards: 70..To:
> <sip:07111234567
> @XX.189.169.66>..From:
> <sip:12 at 192.168.178.26>;tag=z9hG4bK84110414..Call-ID:
> 129926972169 at 192.168.178.26..CSeq: 1 INVITE..Contact: <sip:12 at 192.168.178.2
> 6>..Expires: 3600..User-Agent: mjsip stack 1.6..Content-Length:
> 154..Content-Type: application/sdp....v=0..o=sip:12 at 192.168.178.26 0 0
> IN IP4 192.168.17
> 8.26..s=Session SIP/SDP..c=IN IP4 192.168.178.26..t=0 0..m=audio 21000
> RTP/AVP 0..a=rtpmap:0 PCMU/8000..
> #
> U XX.189.169.66:5060 -> XX.199.123.185:5060
> SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
> 192.168.178.26:5060;branch=z9hG4bK98099;received=XX.199.123.185;rport=5060..From:
> <sip:12 at 192.168.178.26>;tag
> =z9hG4bK84110414..To:
> <sip:07111234567 at XX.189.169.66>;tag=as76b40635..Call-ID:
> 129926972169 at 192.168.178.26..CSeq: 1 INVITE..User-Agent: Asterisk PBX 1
> .6.0.9..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY..Supported: replaces, timer..WWW-Authenticate: Digest
> algorithm=MD5, realm="a
> sterisk", nonce="59645374"..Content-Length: 0....
> #
> U XX.199.123.185:5060 -> XX.189.169.66:5060
> INVITE sip:07111234567 at XX.189.169.66 SIP/2.0..Via: SIP/2.0/UDP
> 192.168.178.26:5060;rport;branch=z9hG4bK98099..Max-Forwards: 70..To:
> <sip:07111234567
> @XX.189.169.66>..From:
> <sip:12 at 192.168.178.26>;tag=z9hG4bK84110414..Call-ID:
> 129926972169 at 192.168.178.26..CSeq: 2 INVITE..Contact: <sip:12 at 192.168.178.2
> 6>..Expires: 3600..User-Agent: mjsip stack 1.6..Authorization: Digest
> username="12", realm="asterisk", nonce="59645374",
> uri="sip:07111234567 at XX.189.1
> 69.66", algorithm=MD5,
> response="b2b86ac54de0f1644da86bc5063e6a21"..Content-Length:
> 154..Content-Type: application/sdp....v=0..o=sip:12 at 192.168.178.26 0
> 0 IN IP4 192.168.178.26..s=Session SIP/SDP..c=IN IP4
> 192.168.178.26..t=0 0..m=audio 21000 RTP/AVP 0..a=rtpmap:0 PCMU/8000..
> #
> U XX.189.169.66:5060 -> XX.199.123.185:5060
> SIP/2.0 491 Request Pending..Via: SIP/2.0/UDP
> 192.168.178.26:5060;branch=z9hG4bK98099;received=XX.199.123.185;rport=5060..From:
> <sip:12 at 192.168.178.26>;
> tag=z9hG4bK84110414..To:
> <sip:07111234567 at XX.189.169.66>;tag=as76b40635..Call-ID:
> 129926972169 at 192.168.178.26..CSeq: 2 INVITE..User-Agent: Asterisk PB
> X 1.6.0.9..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY..Supported: replaces, timer..Content-Length: 0....
> #
>
>
> --
> _____________________________________________________________________
> -- 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
--
Für weitere Fragen stehen wir gerne unter voip at sil.at oder
059944 - 2440 zur Verfügung.
Mit freundlichen Grüssen
--
Stefan Schmidt
Teamleiter VOIP // voip at sil.at // Tel 059944-2440//
-------------------------------------------------
SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 //
A-1160 Wien // Fax 059944-9000 // www.sil.at //
-------------------------------------------------
More information about the asterisk-users
mailing list