We have a new SIP trunk we are trying to implement in our Main Site that Verizon has seperated into two logical trunks.
Trunk 1 = 120 channels for customer IPCC IVR
trunk 2 = 200 channels for our company IPCC
All inbound SIP calls are g729.
Currently the existing system has the Main Site calls on all extensions as g711.
We have a 3845 Gateway with x4 64 DSP SIMMS installed.
When the system was designed, the idea was that only the 120 channels for would need to be transcoded from g729 to g711 for the IVR system because it is a voice banking system. It was thought the these calls needed to be g711 and not compressed locally once inside the building.
Now that we are testing, we are running out of DSP resources.
The Voice contractor is saying that they cannot get this to work this way.
They cannot get it to work with the Main Site extensions at g729, one of the symptoms is that we hear no ring back on out bound calls (everything else seems to work fine).
They are saying that all inbound and outbound calls int the Main site have to be transcoded to g711.
Should we be able to have call Manager handle the inbound SIP calls at g729 and do the IVR processing at g711?
Also, is there an appliance (if we need) that can provide DSP resources, rather than creating a DSP farm with a bunch of different routers?