bridge to gateway overwrites "effective caller id" with "username"
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
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:
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
Maybe it can help by following this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012083.htm... 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@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@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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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:
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
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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> 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@sip.myisp.at <sip%3A%2B4312345678100@sip.myisp.at> ;transport=udp>;tag=U6yQUSta2c2Xg. i get: From: "Desk Phone" <sip:p00xxxx.myisp@sip.myisp.at <sip%3Ap00xxxx.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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
wow, now that was fast :-) Cheers for all replies, setting the caller-id-in-from-parameter was sufficient! regards Christian
participants (5)
-
Anthony Minessale
-
Brian West
-
Christian Benke
-
dujinfang
-
Mathieu Rene