problem with my AA

[coucoumama]

hi all

- i have a gsm gateway connected to my CME router , so i configure all incomming call from outside to 4 SIM on my gsm gateway forwarded to my Auto attend number also my FXO port .

so , the problem when a call join my AA customized from the gsm gateway  , the customer for example asked to press 1 to join the technical team but when she's type it nothing produced and  the prompt keep playing , but from FXO port it work . so i ask for a solution where i think is related to DTMF-relay i think ?

[coucoumama]

can anyone help or have an issue like this ?

[William Bell]

Ben,

How is the GSM gateway connected to the CME router? Is it IP? TDM?

If it is IP then which signaling protocol? SIP? H323?

I assume that you are using BACD on your CME. If that is the case AND you are doing SIP or H323 between the CME and the GSM gateway then you will need to address a few things.

1. For BACD you should be using a voip dial-peer that may look something like:

dial-peer voice 81020 voip

service app-b-acd-aa

destination-pattern 4000    <<< this is your pilot DN for your AA app

session target ipv4:

incoming called-number 4000

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

2. You would need to allow IP-to-IP communication:

voice service voip

allow h323 to h323   <<< if GSM gw is H323

allow sip to h323   <<< if GSM gw is SIP  and also if you are using CUE (SIP integrated)

allow h323 to sip   <<< if I have s to h, I always add h to s

3. DTMF on your GSM voip dial-peer

You have to determine what is supported by the GSM gateway and then provision the voip dial-peer on your CME accordingly. You can provision support for multiple DTMF types as well.

Side note: I have not worked with GSM GW before, so I am speaking from CME PoV only.

HTH.

-Bill (@ucguerrilla)

[mhassani]

Hi

You are right? It is a DTMF issue, what type of connection do you have between you CME and the GSM gateway



Sent from Cisco Technical Support iPad App

[coucoumama]

the GSm gateway is a h323 with ip adress connected to Gb interface to the CME and the type of dtmf on the gsm gateway is alphanumeric , signal and SIP

[coucoumama]

the gsm gtateway is 2n officeroute

[William Bell]

Ben,

OK then you will need a voip dial-peer that your GSM gw can hit and that specifies the correct DTMF-Relay settings.

dial-p v 81020 voip

incoming called

no vad

dtmf-r h245-a

codec

[coucoumama]

is not working when i put the 1 or 2 nothing produced the prompt still play, that's my config

dial-peer voice 102 voip

description Unity Express - VoiceMail

service app

destination-pattern 999

session protocol sipv2

session target ipv4:10.1.1.2

dtmf-relay rtp-nte

no dtmf-interworking

codec g711ulaw

no vad

[coucoumama]

that's a capture formy gsm gw config