Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"
wow, now that was fast :-) Cheers for all replies, setting the caller-id-in-from-parameter was sufficient! regards 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
Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"
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
On Wed, Mar 11, 2009 at 12:07 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,
> ${caller_id_number} is a 3-digit extension, e.g.: 100):
>
> data="effective_caller_id_number=+4312345678${caller_id_number}"/>
> 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"
>
> ;transport=udp>;tag=U6yQUSta2c2Xg.
> i get:
> From: "Desk Phone"
>
> ;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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> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
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googletalk:conf+...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"
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 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,
>> ${caller_id_number} is a 3-digit extension, e.g.: 100):
>>
>> > data="effective_caller_id_number=+4312345678${caller_id_number}"/>
>> > 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"
>> ;tag=U6yQUSta2c2Xg.
>> i get:
>> From: "Desk Phone"
>> ;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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> 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
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Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"
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,
> ${caller_id_number} is a 3-digit extension, e.g.: 100):
>
> data="effective_caller_id_number=+4312345678${caller_id_number}"/>
> data="sofia/gateway/sip.myisp.at/${destination_number}"/>
>
> I'd expect with this dialplan the effective_caller_id would be in the
> "From:"-section of the INVITE, but it seems after the bridge it is
> overwritten with the gateway-username i've defined in the
> gateway-configuration in sip_profiles/external/.
>
> So instead of:
> From: "Desk Phone"
> ;tag=U6yQUSta2c2Xg.
> i get:
> From: "Desk Phone"
> ;tag=U6yQUSta2c2Xg.
> in the INVITE towards the sip-trunk.
>
> I may not have grasped yet how proper debugging with freeswitch works,
> however, in the console the last action i see, before the bridge to
> sofia/external is created, is the setting of the effective-caller-
> id, as
> expected(Do you want to see the whole output?).
>
> I guess i don't necessarily need to register with the provider, as
> they
> have configured the trunk for my ip-adress and i have theirs in
> the ACL(inbound calls work flawless with the head-number+extension),
> so
> maybe the registration is the reason why freeswitch does that
> automatically?
>
> It's probably a little issue, but i don't have the overview yet to
> understand how this happens, maybe someone can point me to the right
> place?
>
> Cheers
> Christian
>
> ___
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"
Try export instead of "set"
/b
On Mar 17, 2009, at 10:23 AM, Christian Benke wrote:
>> > data="effective_caller_id_number=+4312345678${caller_id_number}"/>
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Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"
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 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,
> ${caller_id_number} is a 3-digit extension, e.g.: 100):
>
> data="effective_caller_id_number=+4312345678${caller_id_number}"/>
> 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"
> ;tag=U6yQUSta2c2Xg.
> i get:
> From: "Desk Phone"
> ;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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[Freeswitch-users] 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):
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"
;tag=U6yQUSta2c2Xg.
i get:
From: "Desk Phone"
;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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