H323 trunk to SIP via ACM - problem with DSP resources

[rgrasz]

Hello all.
Scenario:
APC is connected via H323 trunk to ACM and ACM routes calls to PSTN via SIP trunk. When APC start dialing (call setup) reserves DSP channel on G450 and keeps till end of call. Problem is that APC can dial 1000 call the same time and we have only 320 DSP channels on G450. Is there any way to push APC or ACM not to use DSP channel during call setup?
This is very problematic and more and more DSP resources needed...

[mlombardi1]

What is APC? DSP is used when going off-hook and can be released between two IP endpoints if the signaling path between the Avaya elements permits shuffling.

[rgrasz]

APC is Avaya Proactive Contact. Shuffling is enabled on ACM side.
Question is why DSP is used when call is in SETUP mode?

[mlombardi1]

Because DSP is required for dial tone when an endpoint goes off hook. The phone cannot generate tones.

[rgrasz]

Dear mlombardi

APC is connected via H323 TRUNK

this is not endpoint and should'nt use DSP till voice path is established - in this scenario with SIP trunk (IP direct is enabled)
For dialing and signalling should use only signallin channel not DSP - there is no audio yet, no dial tone and busy should be send back in signalling channel.

I agree that phone is different story and i have made test when phone calls in the same way - first use DSP (ofcourse for dial tone) but when call is established DSP was released. Phone H323 avaya type 9670 and the same SIP trunk.