[Asterisk-Users] Asterisk as a Gateway

[pdhales at optusnet.com.au]

Just a quick one - did you do 'make samples' as part of installing asterisk?

That would have given you something to work with, at least.
(all of the files are in the configs folder in the Asterisk src folder if
you want to peruse them)

PaulH

----- Original Message ----- 
From: "Nitesh Divecha" <nitesh at vipernetworks.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Friday, December 30, 2005 9:02 AM
Subject: Re: [Asterisk-Users] Asterisk as a Gateway


> Thanks James,
>
> That should help to start my project.... Thanks a million...
>
> I will keep on updating..
>
> And thanks to all for the inputs....
>
> Thanks,
> Neal
>
>
> On Dec 29, 2005, at 6:39 AM, James Sizemore wrote:
>
> > Nitesh Divecha wrote:
> > > Are there any examples of dial plans? Like how to make the default
> > > context?
> > >
> > > I just need a kick start on the config part, as I am really
> > struggling
> > > on routing the calls.
> > >
> >
> >
> > Here is a very very simple example using a PRI. You will need more
> > error routing in a real dial plan:
> >
> > extensions.conf:
> > [general]
> > static=yes
> > writeprotect=no
> > country=us
> >
> > [local]
> > include => default
> >
> > [globals]
> > TRUNK=Zap/g1
> > LDTRUNK=Zap/g2
> >
> > [trunk]
> > ;Long distance pstn
> > exten => _1NXXNXXXXXX,1,Dial(${LDTRUNK}/${EXTEN})
> > exten => _1NXXNXXXXXX,2,Hangup
> >
> > ;pstn
> > exten => _X.,1,Dial(${TRUNK}/${EXTEN})
> > exten => _X.,2,Hangup
> >
> > [default-out]
> > ;This is where you sent trusted calls from sip.conf out to pstn
> > include => trunk
> >
> > [default]
> > ;you send incoming pstn calls here as well as untrusted voip calls.
> > ;here you would route call to local numbers you own via enum or
> > static.
> > exten => 6153247060,1,Wait(2)   ; you need to wait
> >                                 ; long enough to get
> >                                 ; CNAM off line
> > ;send incoming call to your register server.
> > exten => 5555554444,2,Dial(SIP/5555554444 at inside-voip.com)
> >
> >
> >
> > sip.conf:
> >
> > [general]
> > bindport = 5060
> > bindaddr = 0.0.0.0
> > context = default   ; non trusted call from sip side go here
> > srvlookup = yes
> > dtmfmode=info
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=g729
> >
> > [trusted]
> > type=friend
> > context=default-out  ; trusted call can go out pstn
> > host=192.168.0.1
> > canreinvite=no
> >
> >
> >
> > zaptel.conf:
> > span=1,1,0,esf,b8zs
> > bchan=1-23
> > dchan=24
> > span=2,1,0,esf,b8zs
> > bchan=25-47
> > dchan=48
> > span=3,1,0,esf,b8zs
> > bchan=49-71
> > dchan=72
> > span=4,1,0,esf,b8zs
> > bchan=73-95
> > dchan=96
> > loadzone = us
> > defaultzone=us
> >
> >
> > zapata.conf:
> > [channels]
> > context=default                ;pstn incoming call go here
> > switchtype=national
> > signalling=pri_cpe
> > toneduration=500
> > usecallerid=yes
> > hidecallerid=no
> > callwaitingcallerid=yes
> > echocancel=yes
> > echocancelwhenbridged=yes
> > echotraining=800
> > rxgain=-1.0
> > txgain=-1.0
> > callerid=asreceived
> > ;
> > group=1
> > channel=>1-23
> > channel=>73-95
> > ;
> > group=2
> > channel=>25-47
> > channel=>49-71
> >
> >
> >
> >
> >
> > _______________________________________________
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> >
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> >
>
> Nitesh Divecha
> VoIP/Network Engineer
> Viper Networks
> 10373 Roselle St. Ste:170
> San Diego, CA. 92121
>
> Phone:  858-452-8737
> Fax:      858-452-8638
> Cell:  1-909-964-5181
> vPhone: 544-416-0067
>
> Email: nitesh at vipernetworks.com
> Web: www.vipernetworks.com
>
> "Your Internet Phone Company"
> A publicly traded Company, OTC: VPER
>
>
> _______________________________________________
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