[asterisk-users] SIP trunk between two Asterisk servers
Vieri
rentorbuy at yahoo.com
Wed May 12 16:26:11 CDT 2010
--- On Wed, 5/12/10, Vardan <hvardan71 at gmail.com> 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<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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> >>
> >
> >
> >
> >
>
>
> --
> _____________________________________________________________________
> -- 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
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