call routing

[DHOUHA BEN HAMZA]

Hi,

I have a H323 voice gateway Vg200 with this following configue:

!

voice-port 1/1/1

input gain 14

output attenuation -6

description connection to PSTN

music-threshold -30

supervisory disconnect anytone

no battery-reversal

!

dial-peer voice 1000 voip

destination-pattern ...

progress_ind progress enable 8

max-redirects 10

session target ipv4:173.16.10.200

dtmf-relay h245-alphanumeric

when I call from PSTN :

PSTN call (( example telephonique N° 1234 ))>>> vg200 (( ring, the FXO cards hang up and lisen a dial tone )) >>> (( I compse the Directory number of the Ip Phone that I hope contact example DN 2020 )) >>> callmanager>>> IP Phone.

My probleme it's to eliminate the ring and the dial tone.

I wish dial the whole number to contact the ip phone :

PSTN call (( example call N° 12342020 ))>> vg200>>> callmanager>>> IP Phone.

thanks .

[mimckee]

Sounds like you need to use the connection plar command under the voice port. I recommend using the opx flavor.

Thank you,

-Mckee

[s-cortes]

Hi,

I think your gateway to PSTN is the VG200 and you wish whenever you call from the PSTN, the call will directly ring your IP Phone. In that case, configure a POTS dial-peer wherein you configure "incoming called-number " command. Include also in that POTS dial-peer "direct-inward dial command.

Hope this will help.

Regards,

Sherwin

[gshonting]

The easiest way I have found is to put "connection plar xxx" on the voice-port. The xxx should be an extension at the site you want the calls to go to. Create a voip dial-peer with detination pattern of xxx pointing to call manager. When someone calls the fxo port, it will pick up on the first ring and connect to extension xxx.

voice-port 2/0

input gain 5

connection plar 2130

dial-peer voice 30 voip

destination-pattern 2130

session target ipv4:192.168.100.11

[DHOUHA BEN HAMZA]

Hi,

I don’t wish route all the calls towards single Ip Phone but towards group of Ip Phones.

Best Regards

[jpstott]

I think what you are looking for is DID (Direct-Inward-Dial). This is not supported on FXO ports. You will need a digital line (T1/E1) to accomplish this. There are analog DID VICs available for 2600/3600 routers; unfortunately not for VG200s (at least according to current documentation).

As an alternative, you might look at an IVR Auto Attendant solution either in the gateway or on the Call Manager server.

[s-cortes]

Hi,

I think DID is the solution to your problem, although you are using analog trunk line. Consult your telco provider if it can provide DID functionality on your analog trunks. There is also documentation in Cisco that discusses DID on analog trunk lines. If your planning to subscribe E1/T1 link then this will greatly help your problem since E1/T1 link is already DID capable.

Your your voice gateway, I think VG200 already supports NM-HDV module. So I don't think you need to change it. Just consult this with Cisco representatives.

Regards,

Sherwin