I used to work with */Elastix before ... so, I reset as an Asterisk newbie.  Here's my input.

As long as FS and Asterisk register to each other,  FS is versatile enough to send any SIP URI to Asterisk and vice-versa.  It's up to */Elastix Inbound extensions (dialplan) to route them to your clients. You can expand the Wiki simple extension-to-extension setup to suit your needs.  Specific problems? Just lay them out here. We'll take a look.

/Nandy

On Wed, Jul 16, 2014 at 7:33 AM, Shaw Xu <shawx8@gmail.com> wrote:
I am newbee of freeswitch.( many year experience on asterisk / Elastix / freepbx) 

I checked freeswitch is great as a soft switch(means works as the VoIP router for many PBXes) , and Asterisk (Elastix / freepbx / PIAF and so on) have more feature as the pbx.

So my idea is using freeswitch as the the main hub ( to SIP trunk, which will provide hundreds DID to many PBXes like asterisk / freepbx/elastix.) and freeswitch will collect all minutes to IMS (sip trunk provider) 

I checked forum, looks the wiki only has documents on FreeSwitch and Asterisk directly connect(dial exten each other) , which my intention is using Elastix / PIAF (on top of freepbx) as sub PBX, and the call routered from FreeSwitch should go through the Elastix inbound Routes rules instead the direct extension as in the wiki https://confluence.freeswitch.org/display/FREESWITCH/Asterisk

system environment is CentOS 5.10 with follow the wiki install guide to install freeswitch.( by the way, the MOH not working after install the MOH sound, because the directory is not correct, I copy to the right location manually, and need start the moh for first time.) 

Any ideas are appreciated.



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