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[benke at inqnet.at]

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@sip.myisp.at;transport=udp>;tag=U6yQUSta2c2Xg.
i get:
From: "Desk Phone"
<sip:p00xxxx.myisp@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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[benke at inqnet.at]

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@inqnet.at> wrote:

[brian at freeswitch.org]

Try export instead of "set"

/b

On Mar 17, 2009, at 10:23 AM, Christian Benke wrote:

[mrene_lists at avgs.ca]

gateways have their username in the from section, callerid is sent out
as remote-party-id or p-asserted-identity.
if you want the from part to have the user you need to set the "caller-
id-in-from" param to "true"

Math

On 11-Mar-09, at 1:07 PM, Christian Benke wrote:

[dujinfang at gmail.com]

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:

[anthony.minessale at g...]

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@inqnet.at (benke@inqnet.at)> wrote:

[benke at inqnet.at]

wow, now that was fast

Cheers for all replies, setting the caller-id-in-from-parameter was
sufficient!

regards
Christian

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