--- 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
> >
> >
> >
> >
> >
> 
> 
> -- 
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