[Asterisk-Users] Asterisk as a Gateway

[James Sizemore]

The line that reads:
exten => 6153247060,1,Wait(2)
should have been:
exten => 5555554444,1,Wait(2)


Nitesh Divecha wrote:
> 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"
> A publicly traded Company, OTC: VPER
> 
> 
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