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 <[email protected]> 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:[email protected];transport=udp>;tag=U6yQUSta2c2Xg. >> i get: >> From: "Desk Phone" >> <sip:[email protected];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 > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
