Please change the peers name in any server. for example: server1: interboxsip1
server2: interboxsip2 Vardan Vieri wrote: > > > --- On Wed, 5/12/10, Vardan<[email protected]> wrote: > >> please show "sip show users" and sip >> show peers" > > SERVER 2: > > sip show users (trimmed to just my sip test trunk): > > Username Secret Accountcode Def.Context > ACL NAT > interboxsip mycontext No > RFC3581 > > sip show peers (also trimmed): > > Name/username Host Dyn Nat ACL Port Status > sipprovider/0000000001 w.x.y.z N 5060 OK (90 ms) > interboxsip 192.168.250.111 5060 Unmonitored > 7503/7503 10.215.146.190 D N A 5060 OK (20 ms) > 7502/7502 10.215.146.203 D N A 5060 OK (20 ms) > 7172/7172 192.168.250.7 D N A 13404 OK (40 ms) > 7166/7166 10.215.146.200 D N A 5060 OK (20 ms) > 7165/7165 10.215.248.12 D N A 5060 OK (1 ms) > 7160/7160 10.215.146.182 D N A 5060 OK (20 ms) > 7137/7137 192.168.250.6 D N A 25967 OK (10 ms) > 7118/7118 192.168.250.10 D N A 14508 OK (1 ms) > 7117/7117 10.215.146.185 D N A 5060 OK (20 ms) > 7114/7114 192.168.250.8 D N A 12342 OK (10 ms) > 7112/7112 192.168.250.31 D N A 19829 OK (10 ms) > 7111/7111 192.168.250.32 D N A 35259 OK (80 ms) > 7109/7109 (Unspecified) D N A 0 UNKNOWN > 7097/7097 10.215.146.164 D N A 5060 OK (20 ms) > > SERVER 1: > > sip show users is identical. > > sip show peers (trimmed): > > Name/username Host Dyn Nat ACL Port Status > sipprovider/0000000001 w.x.y.z N 5060 OK (79 ms) > interboxsip 192.168.250.112 5060 Unmonitored > >> >> 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? 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