Dial-Peer and Voice Translation rule

[thorsten-gadau]

Dear all,

I am stucked with dial peers and voice translation rules. I have a Call Manager which has a route pattern 93.!  and this is sent to a Cisco voice router without stripping off 93.

The voice router then should route out all calls coming in with 93 to a specific analogue port. Also the 93 needs to be stripped off and a # needs to be added at the end of the dial string, cause the other end expects a #.

So if I dial 93004912345 the router should send out 004912345# on port 0/0/1. Not sure where to put translation-profile outgoing command, either on the dial peer or n the port.

This is my current dial peer and my voice translation rule ( without adding the # ) 

voice translation-rule 2
rule 1 /.*/ //

voice translation-profile OUTGOING-FBB
translate calling 2

dial-peer voice 112 pots
description ***** Dial via FBB 2 (fxo) *****
preference 2
destination-pattern 93T
progress_ind setup enable 3
progress_ind alert enable 8
port 0/0/1
forward-digits all

voice-port 0/0/1

translation-profile outgoing OUTGOING-FBB
no vad
no comfort-noise
cptone DE
impedance complex2
description ***** Dialout FBB02 (FXO) *****
station-id name FBB02
caller-id enable

I do not know if this is the best way to achieve it, so any other comments how to do it in a better way are welcome.

Cheers

Thorsten

[Vivek Batra]

Couple of points;

1. Please ensure you have inbound dial peer matching all calls from call manager.

2. Although you've applied translation profile but that is not manipulating any digit. By the way if you need to apply translation profiles, you want it on called number however you've applied to calling number.

3. Since outbound dial peer is associated with POTS, explicit matched digits will be stripped automatically however you should remove forward digits all command.

4. If during testing you found that # is not being dialed by IOS, that might be because # being used as default terminating digit. If that is the case, you may need to change terminating digit in gateway to say *. I will double check this.

- Vivek

[thorsten-gadau]

Dear Vivek,

thanks for your reply.

I need the translation profile only for outgoing calls to the PSTN, so I guess the way I have configured it is correct? 

Inbound calls from PSTN over Voice Router to Call Manager are working, thats why I did not post the dial peer and translation profile.

So from my understanding the destination pattern 93T is stripping off the 93 from any dialed number. Then the translation rule takes over, is that correct?

Do you think this translation rule is doing what I need, adding a # to the dialed number?

voice translation-rule 11
Rule 1 /^\(.*\)/ /\1#/

Cheers

Thorsten

[Vivek Batra]

No, translation profile you shared before doesn't seems good as it is applied to calling number whereas you want to manipulate called number. 

Called number by the way converted to desired number automatically because of default POTS dial peer behavior.

Now with respect to #, I would suggest you to test without applying any translation profile and share the result else I will confirm as soon as I get system access to test the same.

- Vivek

[Deepak Mehta]

Yes translation rule is correct and will add #  for every number at the end however you need to apply this as

translate called 2

Then apply the profile outgoing direction on the dial-peer.

[thorsten-gadau]

Thx for the answers, but one thing is confusing me.

The calls flow from CUCM to the Voice router to PSTN, so thats a calling statement.

If I want to dial in from PSTN over the Voice router to the CUCM then it is a called statement or am I totally wrong?

[Deepak Mehta]

Calling is used for translating calling number ANI of the caller.

either coming from PSTN -router - CUCM or from CUCM-router-PSTN.

Called number is used for translating called/dialed  number

either coming from PSTN -router - CUCM or from CUCM-router-PSTN.

[thorsten-gadau]

Ok now I am totally stucked :D

I am also having a SIP connection to a provider on that router and it does exactly what it should. Call from PSTN is forwarded to CUCM and dialed out from CUCM is also working:

voice translation-rule 2
rule 1 /.*/ //

voice translation-rule 99
rule 1 /44238168.*/ /6200/

voice translation-profile INCOMING-SIP-VSAT
translate called 99

voice translation-profile OUTGOING-SIP-VSAT
translate calling 2

dial-peer voice 1001 voip
translation-profile incoming INCOMING-SIP-VSAT
translation-profile outgoing OUTGOING-SIP-VSAT
destination-pattern T
session protocol sipv2
session target dns:sip.myprovider.com
session transport udp
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
fax protocol pass-through g711ulaw
no vad

Is that just luck or is this correct? Sorry but this is really confusing me at the moment :(

So to make it work like I need to have it I need to implement this code here:

voice translation-rule 11
Rule 1 /^\(.*\)/ /\1#/

voice translation-profile OUTGOING-FBB
translate called 11

voice-port 0/0/1
bearer-cap 3100Hz
translation-profile outgoing OUTGOING-FBB
no vad
connection plar opx 6200
description ***** Dialout FBB02 (FXO) *****
station-id name FBB02

dial-peer voice 112 pots
description ***** Dial via FBB 2 (fxo) *****
preference 2
destination-pattern 93T
progress_ind setup enable 3
progress_ind alert enable 8
port 0/0/1

Then calls coming from CUCM into the Voice router flow over port 0/0/1, stripped off the 93 and puts a # at the end of the dialed numbers?

Thx for all the help, much appreciated!!

Cheers

Thorsten

[Vivek Batra]

Hi Thorsten,

I suggest you to share full config to suggest you accordingly.

- Vivek

[Vivek Batra]

Your understanding seems bit wrong. Calling means who is making a call viz ANI irrespective of the call direction and called means to whom call is being made viz DNIS again irrespective of the call direction.

By the way on FXO circuit, you can not manipulate calling number. That can be provisioned only on digital circuit and SIP from PSTN perspective.

-Vivek

[Deepak Mehta]

As per your translation rule 2 you are translating any number to null .

Also because it is defined as translate calling it will affect the ANI for every calls and will translate any ANI to null.So i am not sure why you are doing it ?

Also destination pattern 93T will automatically strip 93  and forward rest of the digits as it is explicit match so this config looks good.