Have you confirmed incomming calls work? if yes, outbound is very simple
ex.. it is an extension your sending back to the pri
exten => _3XX,1,Dial(DAHDI/g1/${EXTEN})
g1 is the group number you setup in you dadhi conf corresponding to the
lines..
Phil
On Wed, Feb 6, 2013 at 11:49 AM, DBC on Asterisk List <[email protected]
> wrote:
> You would need to write 2 dialplan contexts:
> #1 - From SIP provider
> - reads from SIP
> - checks chan availability on the PRI
> - tempfail cust message & congestion - or -
> - dials the PRI
> ** If the legacy PBX picks lines bottom-up (channel 1->23) the dial
> DAHDI/G0 to pick from top-down. If legacy PBX picks line top-down, use
> DAHDI/g0 to pick bottom-up.
>
> #2 - From Legacy PBX
> - read from DAHDI
> - checks chan avail on SIP
> - tempfail cust message & congestion - or -
> - dials SIP
>
> Assuming you aren't modifying the legacy PBX, you now look like the telco
> master. You also have to accept what the telco would in your area - 3 digit
> services, 10-digit dial, 10+ LD & international, etc.
>
> Make sure you check chan availability because as you grow (away from the
> legacy pbx) you won't have 1:1 relationship of trunks-in vs trunks-out. For
> example you start adding SIP trunks/providers, have private trunks to other
> offices, start adding/migrating users from the legacy PBX to Asterisk, etc.
>
> Depending on the situation you could get clever and not answer the call
> while transferring. Depends on the business environment, customer
> use-cases, connect charges from SIP provider, etc. Only you can answer
> which is the right way to approach that.
>
> -dbc
>
>
> On 06/02/2013 10:46 AM, Dean Yorke wrote:
>
>> Hi All,
>>
>> Hope everything is going well.
>>
>> I am looking for some direction on a routing/config issue.
>>
>> Here is what I have.
>>
>> SIP provider connected to an asterisk 1.11 box with a sangoma PRI that is
>> connected with a crossover cable to a legacy PBX.
>>
>> I am using freepbx 2.10.
>>
>> I am trying to understand how I can route a sip call through asterisk/pri
>> to legacy pbx for both inbound and outbound calls. I think i understand
>> the outbound fine it's the inbound i am having questions about.
>>
>> The PRI is configured as from-internal context. And it is synced with
>> the legacy pri port.
>>
>> Suggestions?
>>
>> thanks for you help.
>>
>>
>>
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