DTMF tones to Fax Server not being sent.

Unanswered Question
Feb 12th, 2008

Greeting's,

I have an ISR gtwy with a mgcp controlled PRI (from PSTN) and an FSX/DID card communicating H.323 to UCM Business Edition 6.0. The FXS handoff is to a Faxpress server, in UCMBE I have a route pattern pointing to the h.323 gtwy. Faxes hit the Faxpress server but are continuously sent to the default Unaddressed box because the DTMF tones are not being sent. I have tried just about everyhting from going all MGCP back to H.323 and making my adjustments on the dial-peers. Is their any know bugs with UCMBE 6.0?

Thx

I have this problem too.
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pcameron Tue, 02/12/2008 - 13:39

MGCP wil not allow digits to be sent out a FXS port, so this won't work. You need to run the FXS via H323 with a config similar to this -

Assuming you want to send digits in the range 1000 - 1999:

dial-peer voice 10 pots

description - connection to fax server

destination-pattern 1...

prefix ,,,,1

The 'prefix ,,,,1' will add a 1 second pause after the voice port detects the answer. Each ',' represents a 250msec delay.

It will then send the original leading digit (1) and the other digits that should match the fax mailbox number. We have mde this work many times on fax servers so give it a try and see if it helps.

It's quite OK to have the PRI under MGCP control and the FXS under H323 on the same device.

hbarrera2 Tue, 02/12/2008 - 14:46

pcameron,

Thanks for the prompt reply, I will try your suggestion right now and let you know!

thx again

hbarrera2 Tue, 02/12/2008 - 15:02

pcameron,

The following is my configurations. I added the voip dial peer since the call leg from CCM to the FXS is voip. Still not sending dtmf though...

dial-peer voice 46981 voip

session target ipv4:10.17.32.32

incoming called-number 4698

dtmf-relay h245-alphanumeric

codec g711ulaw

fax protocol pass-through g711ulaw

no vad

!

dial-peer voice 4698 pots

destination-pattern 4698

port 0/1/2

prefix ,,,,4698

!

pcameron Tue, 02/12/2008 - 15:27

it should be ... do a debug voip vtsp all and debug vpm signal and log/paste these results here.

Otherwise, plug a telephone handset into the port and call through to it - when you answer you should here the tones. Please confirm if this happens.

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