[asterisk-users] SIP trunk between two Asterisk servers
Vieri
rentorbuy at yahoo.com
Wed May 12 12:25:55 CDT 2010
--- On Wed, 5/12/10, Vardan <hvardan71 at gmail.com> 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<klitzing at 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:3666 at 192.168.250.111 SIP/2.0
> > Via: SIP/2.0/UDP
> 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
> > From:
> "device"<sip:4053 at 192.168.250.112>;tag=as18a568d6
> > To:<sip:3666 at 192.168.250.111>
> > Contact:<sip:4053 at 192.168.250.112>
> > Call-ID:
> 328617546726e5d430538e80617716e1 at 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 -
> 328617546726e5d430538e80617716e1 at 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:4053 at 192.168.250.112>;tag=as18a568d6
> > To:<sip:3666 at 192.168.250.111>;tag=as57a19dac
> > Call-ID:
> 328617546726e5d430538e80617716e1 at 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
> '328617546726e5d430538e80617716e1 at 192.168.250.112' in 15000
> ms
> > Found user '4053'
> >
> > <-- SIP read from 192.168.250.112:5060:
> > ACK sip:3666 at 192.168.250.111 SIP/2.0
> > Via: SIP/2.0/UDP
> 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
> > From:
> "device"<sip:4053 at 192.168.250.112>;tag=as18a568d6
> > To:<sip:3666 at 192.168.250.111>;tag=as57a19dac
> > Contact:<sip:4053 at 192.168.250.112>
> > Call-ID:
> 328617546726e5d430538e80617716e1 at 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
> >
> >
> >
> >
> >
>
>
> --
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