[Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"

[dujinfang]

Maybe it can help by following this thread

http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012083.html


On Mar 17, 2009, at 11:23 PM, Christian Benke wrote:

> Hi!
>
> Is this not possible with registration at a gateway or is there a  
> other
> reason why i didn't get any responses on this question?
>
> Regards
> Christian
>
> On Wed, 11 Mar 2009 18:07:42 +0100
> Christian Benke <benke at inqnet.at> wrote:
>
>> Hi!
>>
>> I've recently started to configure a freeswitch for our new office  
>> pbx
>> and so far i like it very much(Coming from asterisk&openser with 2
>> years experience at a ITSP. Openser was nice but i didn't like
>> asterisk for several reasons, so i searched for a more stable and
>> cleaner alternative. Freeswitch looks _very_ promising and i'd wished
>> i could use it for more difficult demands than a simple
>> office-pbx ;-)).
>>
>> So far i had little trouble(Though our installation doesn't require
>> much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP.
>>
>> The only issue i have not resolved yet is setting the outgoing
>> DID("head"-number + extension, e.g. +4312345678 + 100).
>>
>> The relevant part of the default.xml looks like this atm(where
>> +4312345678 is our "head"-phone-number without the extensions,
>> ${caller_id_number} is a 3-digit extension, e.g.: 100):
>>
>> <anti-action application="set"
>> data="effective_caller_id_number=+4312345678${caller_id_number}"/>
>> <anti-action application="bridge"
>> data="sofia/gateway/sip.myisp.at/${destination_number}"/>
>>
>> I'd expect with this dialplan the effective_caller_id would be in the
>> "From:"-section of the INVITE, but it seems after the bridge it is
>> overwritten with the gateway-username i've defined in the
>> gateway-configuration in sip_profiles/external/.
>>
>> So instead of:
>> From: "Desk Phone"
>> <sip:+4312345678100 at sip.myisp.at;transport=udp>;tag=U6yQUSta2c2Xg.
>> i get:
>> From: "Desk Phone"
>> <sip:p00xxxx.myisp at sip.myisp.at;transport=udp>;tag=U6yQUSta2c2Xg.
>> in the INVITE towards the sip-trunk.
>>
>> I may not have grasped yet how proper debugging with freeswitch  
>> works,
>> however, in the console the last action i see, before the bridge to
>> sofia/external is created, is the setting of the effective-caller-id,
>> as expected(Do you want to see the whole output?).
>>
>> I guess i don't necessarily need to register with the provider, as
>> they have configured the trunk for my ip-adress and i have theirs in
>> the ACL(inbound calls work flawless with the head-number+extension),
>> so maybe the registration is the reason why freeswitch does that
>> automatically?
>>
>> It's probably a little issue, but i don't have the overview yet to
>> understand how this happens, maybe someone can point me to the right
>> place?
>>
>> Cheers
>> Christian
>
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