CPN issue

[prempreet]

Hi Team,

Good day..!

We have two requirements from client.

1) As per client requriement, one extension should have one CPN. For example, if we dial from Extension1 then one CPN will display on the customer's phone. If we dial from Extension 2, then another CPN will display on the customer's phone

2) For this requirment, i have configured CPN extension wise. Due to this, during inernal calls, CPN number is also displaying on the extension wise. If there is any internal missed calls, we couldn't callback. The call is disconnecting

Anyone, please help me to slove this

Regards
Prem

[bublepaw]

In general SIPS has 3 different options to change number presented to other party:

cpn - changes user part of From header
display-name - changes display-name of From header
p-asserted-identity - changes or adds pai header

To solve Your issue You have two options:

a) configure gateway and phones to use different fields/headers to present number - for example configure gateway to present number from pai header and configure phone to present number from user part or gateway can take number presentation from user part and phone can take it from display-name etc. Second part of configuration is to use options from above to add proper presentation for each device. Some phones do not have option to choose what to present but for example always uses pai header if it is present.

b) use dialplan to change number stored on the phone (taken from CPN) into internal number

[prempreet]

Can You please explain the configuration actually i am not able to understand the solution you gave.

[prempreet]

Sorry, actually i am new to Genesys & i couldn't understand.your inputs. And i read the sipserver deployment document for p-asserted-identity option & for assigning PAI header. But we couldn't understand.

It will be great help, if you can explain little bit clearly like,

1) how to configure the gateway to take number presentation from user part or PAI header?

  OR

2) how to configure the phone to take number from display name?

Thanks in advance.

[Kubig]

It depends on used gateway type,you have to set it on GW side.Second solution which bublepaw gave to you is about genesys dial-plan,one of features of T-Server when you can create NumberingPlan as on a PBX,but my suggestion is to set these features on GW side.

[prempreet]

Thanks for input.

We are working with Audio code(Mediant 1K) engineer. Will test & revert back.

[bublepaw]

Sorry for delay - I had to do some testing. On Audiocodes You have option called "Use Display Name as
Source Number" - set it to true and on all phones in Genesys set display-name to specify what will be presented by the gateway. On Yealink phone there is option called "Caller ID Header" when from dropdown You can select which field is used for presentation. On Polycom it is not possible to configure (at least I didn't find option) but at least Polycom has priority - it first looks and PAI, then dispaly name and at the end at user part.

If You have Audiocodes/Polycom combination:
- set Use Display Name as Source Number to true on Audiocodes
- in genesys configure display-name=number presented to external world, p-asserted-identity = internal number

If You have Audicodes/Yealink combination:
- set Use Display Name as Source Number to true on Audiocodes
- in genesys configure display-name=number presented to external world, p-asserted-identity = internal number
- in Yealink set Caller Id header to value 2 (PAI-FROM)

[ramesh]

Sorry for the late reply... I was working with AudioCode engineer for this issue. The configuration which you suggested is working, if i dial the number from IWS.

But if i dial from Yealink T32(sipphone), the same problem is coming. In our setup, the agents used to dial from Yealink instead of IWS.

Regards
Ramesh

[cavagnaro]

1PCC is a little bit different than 3PCC (IWS), check DN config...not 100% sure will work. Also compare SIP/TLIB Traces from both scenarios to understand what is going on