please show "sip show users" and sip show peers" vardan
Vieri wrote: > > > --- On Wed, 5/12/10, Philipp von > Klitzing<[email protected]> wrote: > >>> <--- SIP read from 192.168.250.111:5060 ---> >>> SIP/2.0 407 Proxy Authentication Required >> >> You need to run the SIP debug on 192.168.250.111 to learn >> more about WHY >> the 407 is issued. Have a close look and you are likely to >> understand it >> right away. >> >> Also: Do not forget the "reload" after applying changes to >> sip.conf. > > I always do a "sip reload" after changes to sip settings. > > Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving > end): > > <-- SIP read from 192.168.250.112:5060: > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport > From: "device"<sip:[email protected]>;tag=as18a568d6 > To:<sip:[email protected]> > Contact:<sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Wed, 12 May 2010 09:20:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > upported: replaces > Content-Type: application/sdp > Content-Length: 270 > > v=0 > o=root 20611 20611 IN IP4 192.168.250.112 > s=session > c=IN IP4 192.168.250.112 > t=0 0 > m=audio 14648 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > --- (14 headers 13 lines) --- > Using INVITE request as basis request - > [email protected] > Sending to 192.168.250.112 : 5060 (NAT) > Reliably Transmitting (NAT) to 192.168.250.112:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 > From: "device"<sip:[email protected]>;tag=as18a568d6 > To:<sip:[email protected]>;tag=as57a19dac > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1327c5b6" > Content-Length: 0 > > > --- > Scheduling destruction of call > '[email protected]' in 15000 ms > Found user '4053' > > <-- SIP read from 192.168.250.112:5060: > ACK sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport > From: "device"<sip:[email protected]>;tag=as18a568d6 > To:<sip:[email protected]>;tag=as57a19dac > Contact:<sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > Can you deduce from this what I'm doing wrong? > > Thanks, > > Vieri > > > > > -- _____________________________________________________________________ -- 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
