--- On Wed, 5/12/10, Vardan <[email protected]> wrote:
> Please change the peers name in any > server. > for example: > server1: > interboxsip1 > > server2: > interboxsip2 If I understand correctly, the peer names can be identical on both servers. What counts is the "host" entry, I guess. But then again, my SIP trunk isn't working so I'll try out your suggestion tomorrow. Thanks, Vieri > > 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? 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 > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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