Adding a Zero Prefix to SIP FROM header value

[PFCCWA]

Does anyone know if it is possible to add a zero(0) prefix to the ANI within the SIP FROM header.
At the moment calls displaying on the SIP End Point do so without the 0.
This end point documentation states this is taken from the FROM header field in the SIP message.
SIP logs show this under the INVITE header:
...
CSeq: 1 INVITE
From: sip:1234567891@prgroup.prodfin.com;tag=80b8d0cf7e8e216a204f2131bd00
...

Using SIP 8.0
Avaya Media Gateway
GVP 8.1.2

Thanks.

[bublepaw]

Hi,

Question is too generic - we need more info:

a) which part of FROM header You want to modify
b) what is exact callflow in which You need header to be modified
c) is it only FROM or also some other fields that needs to be modified

[peters]

Hi, i'm dealing with same problem. I need to rewrite the Contact: field. I used the Tlib-to-SIP mapping via SIP_HEADERS in IRD strategy . I ended up with two Contact fields, it was not rewritten, even when i used ExtensionUpdate.

The reason is to forward incoming call to external party via Troute (ISDN provider doesn't allow to pass the originating ANI to forwarded destination).

[Kubig]

I thought this could be solved by CCP(Call Control Platform) from VP8.1

[PFCCWA]

[quote author=bublepaw link=topic=7432.msg31794#msg31794 date=1348007796]
Hi,

Question is too generic - we need more info:

a) which part of FROM header You want to modify
b) what is exact callflow in which You need header to be modified
c) is it only FROM or also some other fields that needs to be modified
[/quote]

I want to modify the ANI, by adding a 0 to the number.  The SIP End Point uses this to display the caller id so at the moment users have to add the 0 manually when calling back.  The part is:
From: sip:[u][b]1234567891[/b][/u]@prgroup.prodfin.com;tag=80b8d0cf7e8e216a204f2131bd00
Im hoping there is way to update so displays as this:
From: sip:[u][b]01234567891[/b][/u]@prgroup.prodfin.com;tag=80b8d0cf7e8e216a204f2131bd00

There is no requirement to change the call flow, just this part of the INVITE (header).
The SIP End Point documentation advises it takes the caller ID from the 'FROM' message.

Thanks.

[peters]

You could create trunk, and add prefix there... This is how we do it usually.

[Adam G.]

A missing zero at the begining of the number is normally a sign that the 'number' has been converted to the datatype Integer. If it was treated as a Varchar then the 0 would probably still be present.

[PFCCWA]

[quote author=peters link=topic=7432.msg31799#msg31799 date=1348047872]
You could create trunk, and add prefix there... This is how we do it usually.
[/quote]

How did you configure this?
Add a prefix option to the TServer section of the inbound Trunk? What value?
I attempted this on our media gateway trunk without success (added section=TServer, option=prefix, value=0)
Also tried a VOIP with service type dial-plan - no luck.

Thanks.

[peters]

Sorry i thought you want to overwrite the outgoing call FROM.
Is the call routed via URS? If so, you could edit the udata ANI. I also tried the dialplan, but it didn't work.
Are u using some Genesys Desktop app? You could do customization there, to attach the 0...

[bublepaw]

Hi,

Part of From header that You want to modify is not accessible to external process outside SIPS so there is no easy way to modify it. But as usual there is another way. You can create dialplan attached to DN's with follwing rule "XXXXXXXXXX => 0${DIGITS}". It should do the trick that when user tries to redial from the phone it will be corectly processed by SIPS.