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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