[Asterisk-Users] Asterisk as a Gateway

[Nitesh Divecha]

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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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"
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