The From: header is not the correct place to place the caller id in SIP yet some providers assume it is.<br>If you add this to your gateway xml config it should fix your problem<br><br><param name="caller-id-in-from" value="true"/><br>
<br><br><div class="gmail_quote">On Wed, Mar 11, 2009 at 12:07 PM, Christian Benke <span dir="ltr"><<a href="mailto:benke@inqnet.at">benke@inqnet.at</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi!<br>
<br>
I've recently started to configure a freeswitch for our new office pbx<br>
and so far i like it very much(Coming from asterisk&openser with 2<br>
years experience at a ITSP. Openser was nice but i didn't like asterisk<br>
for several reasons, so i searched for a more stable and cleaner<br>
alternative. Freeswitch looks _very_ promising and i'd wished i could<br>
use it for more difficult demands than a simple office-pbx ;-)).<br>
<br>
So far i had little trouble(Though our installation doesn't require<br>
much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP.<br>
<br>
The only issue i have not resolved yet is setting the outgoing<br>
DID("head"-number + extension, e.g. +4312345678 + 100).<br>
<br>
The relevant part of the default.xml looks like this atm(where<br>
+4312345678 is our "head"-phone-number without the extensions,<br>
${caller_id_number} is a 3-digit extension, e.g.: 100):<br>
<br>
<anti-action application="set"<br>
data="effective_caller_id_number=+4312345678${caller_id_number}"/><br>
<anti-action application="bridge"<br>
data="sofia/gateway/<a href="http://sip.myisp.at/$%7Bdestination_number%7D" target="_blank">sip.myisp.at/${destination_number}</a>"/><br>
<br>
I'd expect with this dialplan the effective_caller_id would be in the<br>
"From:"-section of the INVITE, but it seems after the bridge it is<br>
overwritten with the gateway-username i've defined in the<br>
gateway-configuration in sip_profiles/external/.<br>
<br>
So instead of:<br>
From: "Desk Phone"<br>
<<a href="mailto:sip%3A%2B4312345678100@sip.myisp.at">sip:+4312345678100@sip.myisp.at</a>;transport=udp>;tag=U6yQUSta2c2Xg.<br>
i get:<br>
From: "Desk Phone"<br>
<<a href="mailto:sip%3Ap00xxxx.myisp@sip.myisp.at">sip:p00xxxx.myisp@sip.myisp.at</a>;transport=udp>;tag=U6yQUSta2c2Xg.<br>
in the INVITE towards the sip-trunk.<br>
<br>
I may not have grasped yet how proper debugging with freeswitch works,<br>
however, in the console the last action i see, before the bridge to<br>
sofia/external is created, is the setting of the effective-caller-id, as<br>
expected(Do you want to see the whole output?).<br>
<br>
I guess i don't necessarily need to register with the provider, as they<br>
have configured the trunk for my ip-adress and i have theirs in<br>
the ACL(inbound calls work flawless with the head-number+extension), so<br>
maybe the registration is the reason why freeswitch does that<br>
automatically?<br>
<br>
It's probably a little issue, but i don't have the overview yet to<br>
understand how this happens, maybe someone can point me to the right<br>
place?<br>
<br>
Cheers<br>
Christian<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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