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

[Anthony Minessale]

The From: header is not the correct place to place the caller id in SIP yet
some providers assume it is.
If you add this to your gateway xml config it should fix your problem

<param name="caller-id-in-from" value="true"/>


On Wed, Mar 11, 2009 at 12:07 PM, 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/${destination_number}<http://sip.myisp.at/$%7Bdestination_number%7D>
> "/>
>
> 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 <sip%3A%2B4312345678100 at sip.myisp.at>
> ;transport=udp>;tag=U6yQUSta2c2Xg.
> i get:
> From: "Desk Phone"
> <sip:p00xxxx.myisp at sip.myisp.at <sip%3Ap00xxxx.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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-- 
Anthony Minessale II

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