On Wed, Feb 3, 2010 at 6:07 PM, Jim Canfield <[email protected]> wrote:
> Hey guys,
>
> I have an interesting situation using X-lite 3.0 soft-phones and the
> 4.0 Beta.  Before I start gathering sip traces, I thought I'd see if
> this is the expected behavior...
>
> 1 -  Created soft-phone users.  Sation01, Station02...
> 2 -  Registered soft-phones manually (i.e. no phone provisioned for
> these users.)
> 3 -  Everything works fine...except outgoing caller ID  is '01' for
> Sation01 and '02' for Station02...etc.
>
> I've tried changing outgoing caller ID on the phone groups and each
> user individually without success.  All the Polycoms work fine.  Is
> this expected behavior when there is no phone assigned to a user?
>
> GW - Patton 4960
> sipXecs - 4.0.4
>

I tried using 'real' extension numbers as Mike suggested with the same
results.  Only now it is sending the extension as the caller id.  I'm
still trying to decide if this is a sipXecs issue or an issue with the
softphone.  Looking at the sip signaling it's obvious the sip URL is
not being changed in sipX.  The sipX from address URL is hitting the
gateway unchanged.

Example:

[email protected] should be changed to [email protected].

My question at this point is at what point is that header rewritten
and what triggers it?  Is it a requirement that the user have a
registered phone in order for the header to be rewritten before being
passed to the gateway?  I guess I could buy a licence for eyebeam and
see if that changes things.

Scott, you are right. The more things move away from the PSTN, the
less relevant caller id routing becomes.  In fact I think it could
potentially convolute the entire process.
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