[asterisk-users] SIP trunk between two Asterisk servers

[Vieri]

--- On Wed, 5/12/10, Vardan <hvardan71 at gmail.com> wrote:

> And "sip show registry"

sip show registry doesn't list anything regarding my "interboxsip" test trunk because I'm trying to setup a straightforward link such as this one described here (without user/password):
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/

The only "sip show registry" entry I have is the one for my external Internet SIP trunk, which is ok.

Thanks for your time.

> 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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