--- On Wed, 5/12/10, Vardan <hvarda...@gmail.com> wrote:
> And "sip show registry" sip show registry doesn't list anything regarding my "interboxsip" test trunk because I'm trying to setup a straightforward link such as this one described here (without user/password): http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/ The only "sip show registry" entry I have is the one for my external Internet SIP trunk, which is ok. Thanks for your time. > Vardan > > Vieri wrote: > > > > > > --- On Wed, 5/12/10, Philipp von > > Klitzing<klitz...@pool.informatik.rwth-aachen.de> > 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:3...@192.168.250.111 SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport > > From: > "device"<sip:4...@192.168.250.112>;tag=as18a568d6 > > To:<sip:3...@192.168.250.111> > > Contact:<sip:4...@192.168.250.112> > > Call-ID: > 328617546726e5d430538e8061771...@192.168.250.112 > > 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 - > 328617546726e5d430538e8061771...@192.168.250.112 > > 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:4...@192.168.250.112>;tag=as18a568d6 > > To:<sip:3...@192.168.250.111>;tag=as57a19dac > > Call-ID: > 328617546726e5d430538e8061771...@192.168.250.112 > > 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 > '328617546726e5d430538e8061771...@192.168.250.112' in 15000 > ms > > Found user '4053' > > > > <-- SIP read from 192.168.250.112:5060: > > ACK sip:3...@192.168.250.111 SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport > > From: > "device"<sip:4...@192.168.250.112>;tag=as18a568d6 > > To:<sip:3...@192.168.250.111>;tag=as57a19dac > > Contact:<sip:4...@192.168.250.112> > > Call-ID: > 328617546726e5d430538e8061771...@192.168.250.112 > > 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 > -- _____________________________________________________________________ -- 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