I have been working on getting telecenter u fully integrated into fusionpbx via sip. Telecenter u is a district wide voip pa/intercom platform that sends audio to its own devices via rstp, but it can accept and redirect its traffic via sip. So far I created a gateway using its local ip and sip port as the proxy, this device does not support any sip auth, so reg is set to false with nothing in the user and pw. I created a outbound route 9 + 4 dial plan to direct the last 4 to the telecenter gateway. I have created 2 different inbounds directing the calls from the telecenter u with leading digits 5 and 6 to two different fusion reg'd devices. All of this works very well.
I have one feature in the telecenter u called "stream to sip". You define some sip phones(phones extensions in fusion) in telecenter, then you can zone these devices and then direct audio events to them. When this is done, the only thing that comes to freeswitch is 0@<ip-addr-of-telecenter>:5060. I am wanting this to be redirected as a page lua *724 in fusionpbx, so that this audio event can also be broadcasted to all of the pbx devices. I have tried a couple methods with inbound routes but they have failed miserably.
I just started doing voip pbx stuff, self taught on the fly and really loving this software. I recently sold a 120 phone deployment and am looking forward to seeing how this system performs in a real world high call volume atmosphere. Any and all help is appreciated.
kb
I have one feature in the telecenter u called "stream to sip". You define some sip phones(phones extensions in fusion) in telecenter, then you can zone these devices and then direct audio events to them. When this is done, the only thing that comes to freeswitch is 0@<ip-addr-of-telecenter>:5060. I am wanting this to be redirected as a page lua *724 in fusionpbx, so that this audio event can also be broadcasted to all of the pbx devices. I have tried a couple methods with inbound routes but they have failed miserably.
I just started doing voip pbx stuff, self taught on the fly and really loving this software. I recently sold a 120 phone deployment and am looking forward to seeing how this system performs in a real world high call volume atmosphere. Any and all help is appreciated.
kb