[Freeswitch-users] Rewriting Remote Party ID

[Peter P GMX]

Ok, I tried it. It works fine on dialthru. SDP header is enhanced. Perfect!

However I have a problem which I already discussed before in this round,
and I am very close to solve it, I think.
Scenario is as follows

    * A callback application which is done through event socket calls
      party A with cid-number 'unknown'
    * Party A enters Digits which represents the number of party B
    * I set the caller-id-no and effective-caller-id-no to the value of
      A and  sip_cid_type=rpid
    * Call is then transferred from A to Party B via the xml dialplan
    * Now I have 2 different variables when I trace the event_socket
          o caller-caller-id-number is 'anonymous' and this is submitted
            in the INVITE message to B
          o variable-caller-id-number is the right number of B, this is
            not part of the Sip message
    * B sees 'unknown' as the number of party A. This is also
      represented in the SIP mssage sent to B
    * I want B to see the right number of party A

My question: What can I do to set <effective-caller-id-no> to
<variable-caller-id-number> in the XML dialplan.

Here is the part of the dialplan where I want so set it:
    <!-- Dial any 10 digit number -->
    <extension name="10 digits">
      <condition field="destination_number" expression="^mp_([1-9]\d{9})$">

       <action application="bridge" data="sofia/external/1$1 at domain.com"/>
      </condition>
    </extension>


Anybody has a solution?

Best regards
Peter




Ken Rice schrieb:
> These should be available any time you are going to process a call thru the
> dialplan and call a bridge on the call
>
>
>   
>> From: Peter P GMX <Prometheus001 at gmx.net>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 10 Mar 2009 23:20:39 +0100
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] Rewriting Remote Party ID
>>
>> Hello,
>>
>> are these variables only available at call setup time or can they be
>> changed during a call, e.g. before a call is being transferred to
>> another destination?
>>
>> Best regards
>> Peter
>>
>> Michael Collins schrieb:
>>     
>>> On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>   
>>>       
>>>> Latest SVN:
>>>>
>>>> Send no extra caller id info:
>>>> {sip_cid_type=none}sofia/default/user at example.com
>>>>
>>>> Send RPID (default)
>>>> {sip_cid_type=rpid}sofia/default/user at example.com
>>>>
>>>> Send P-XXX-Identity
>>>> {sip_cid_type=pid}sofia/default/user at example.com
>>>>
>>>> Send RPID with chosen content
>>>> {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_num
>>>> ber=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul
>>>> tuser at example.com
>>>>     
>>>>         
>>> FYI, I added this info to the channel variables page:
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type
>>>
>>> -MC
>>>
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>
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