Thanks, but my provider send me a 10 digits number and I want to transalte it in a number like 2..

Just configure the 10 digits number as secondary in the corresponding ephone-dn. You can also configure sip-ua and it will register this number (no-reg primary) to the ITSP.

Then, "debug ccsim message" is your friend and will tell you what the ITSP is sending, and why is rejected.

You can create a translation rule like the following:

Lets say your provider sent you:

919 991 5454

voice translation-rule 100

rule 1 /^919....\(...\)$/ /\1/

voice translation-profile TOEXT

translate called 100

dial-peer voice 2000 voip

translation-profile incoming TOSIP

session protocol sipv2

dtmf-relay rtp-nte

codec g711ulaw

This should do it.

Number converted will be 454

or use:

dialplan-pattern command

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_administration_guide_chapter09186a00801f12a8.html#wp2261346

Again, the voice translation-rule, although a very powerful tool, is not necessary in this case.

The secondary number in ephone-dn, is instead what the CME architecture gives you to easily route call coming in as "pots" numbers.

Here you can see part of my configuration. It still don't work...

The dial-peer 3 should the one which is used for incoming calls ...

dial-peer voice 1 voip

translation-profile outgoing OutgoingProfile

destination-pattern T

session protocol sipv2

session target ipv4:213.161.201.200

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 2 voip

destination-pattern 8..

session protocol sipv2

session target ipv4:10.10.0.33

session transport tcp

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 3 voip

session protocol sipv2

incoming called-number 0877192763

dtmf-relay rtp-nte

codec g711alaw

(...)

sip-ua

authentication username login password ****

sip-server ipv4:213.161.201.200

notify telephone-event max-duration 500

(...)

ephone-dn 1 dual-line

number 200 secondary 0877192763 no-reg primary

can you post output of "debug ccsip message" ? You will need of course "term mon" to see it on a telnet connection.

Dial-peer 3, being an incoming only, does not even need "session protocol", but I'm not sure if this is the problem.

I get no messages from a debug command using a console connection.

I will phone the provider to see if he can see something wrong ...

You must be able to see debug output, that is key to working with routers. Telnet to the router, and type "terminal monitor" before or after the debug commands.

I can show debug, but only during outgoing calls... nothings appears during an incomming call. That's why I'm trying to find a solution directly with my provider.

I'll let you know

Possibly you need to register with ITSP ro receive incoming calls. Create a "voice register dn" with the number that ITSP wants to be registed and a register server in sip-ua.

Woops, I see that your local phones are SCCP. Then no "voice register dn" is necessary, just "registrar" under "sip-ua".

Thanks for help !

Everything is working now.

Another question..

If I configure my cisco like you said, the ephone-dn registers the phone number with the secondary line.

Is possible to register with a dial-peer or something else to use hunt-groups ?