Dial Plan for OCS Calling List records?

[PFCCWA]

Hello,

Is it possible to apply a dial plan for ocs calling list records?
We have some records that contain an international prefix, which have call outcome of SIT Intercept.

If I create dial plan VOIP and add as annex option in trunk group, the same call result occurs.
I know the dial plan works because if i do the same for extension DN and use GAD to dial manually, the number is successfully translated.
option used is (section=TServer, option=dial-plan, value=[name of dial plan].

Am I doing this correctly?
We use SIP 8.1, GVP MCP 8.1.5 for CPD, OCS 8.1.3.

thanks.

[Fra]

It sounds like the right approach to me..the dial plan should be applied on the resource dialling, and here that's the trunk group.
The call result you get when a call is barred is strictly related to the way you implement the dial plan.

Fra

[PFCCWA]

There is no bar because international numbers do work, just any numbers with a prefix of the local international code.
I think this has not been allowed at pstn level due to having too many digits so need a translation at the genesys side .
if this is not possible to do at calling list level then i could try it at the avaya end.

thanks.

[René]

Hi PFCCWA,

I have tested it myself but it should work starting SIP Server release 8.1.101.03. Below is description extracted from release notes.

[quote]SIP Server can now apply Dial Plan rules to outbound DNs provided in the OtherDN attribute of TMakePredictiveCall when a dial plan is assigned to the DN initiating the call. For this feature to work, a designated Dial Plan can now be assigned to DNs of type Trunk Group and Routing Point.

For TMakePredictiveCall scenarios, SIP Server supports the following parameters in the dial-plan rule:

- type (with digits and reject values)
- calltype (with outbound values)
- clir
- privilege

This feature applies to outbound calls in Proactive, ASM, and Transfer modes. It is not supported for Dial Plans provided by SIP Feature Server.[/quote]

R.

[PFCCWA]

thank you, worked upon update to sip server