[Freeswitch-users] sip_cid_type settings
Anthony Minessale
anthony.minessale at gmail.com
Thu May 27 06:44:53 PDT 2010
ya, wrong param,
This is the right one:
<!-- Remote-Party-ID header -->
<param name="caller-id-type" value="rpid"/>
or
<!-- P-*-Identity family of headers -->
<param name="caller-id-type" value="pid"/>
or
<param name="caller-id-type" value="none"/>
On Thu, May 27, 2010 at 6:40 AM, David Ponzone <david.ponzone at gmail.com>wrote:
> Frank,
>
> you are totally right.
> I was probably tired when I sent that :)
>
> I think too adding sip-cid-type as a gateway param would be logical and
> nice.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
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>
> Le 27/05/2010 à 13:05, Frank @ Impact a écrit :
>
> Anthoney,
> Maybe you replied to the wrong thread or I am missing the connection
> betweensip_cid_type and dtmf-type. Is there a connection? However, I
> also found that same example in about the same 10 seconds of time. I just
> was unsure how it related tosip_cid_type. Is there a connection?
>
> David,
> The accountcode is a nice idea. But I do not see it in the sample
> sip_profiles as an available option to set. If that variable is set in
> the sip_profile, will it carry through and be available in the dialplan? It
> seems that at the time of the dialplan, FS has not yet determined which
> route it is going to take and so does not yet know which sip_profile is
> going to be used and does not yet have those variables available to the
> call. Am I understanding this wrong?
>
> -----Original Message-----
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>
> ] *On Behalf Of *Anthony Minessale
> *Sent:* Tuesday, May 25, 2010 11:01 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] sip_cid_type settings
>
> well i was not sure so I looked in the example config and searched for dtmf
> in under 10 seconds I was able to do the following from the build root.
>
> cat conf/sip_profiles/internal.xml | grep dtmf
> <param name="dtmf-duration" value="2000"/>
> <!--<param name="dtmf-type" value="info"/>-->
>
>
>
>
> 2010/5/25 David Ponzone <david.ponzone at gmail.com>
> Frank,
>
> as far as I know, there is not.
> You can probably simulate that by using a custom variable in the gateway
> profile, like accountcode, and then matching that at the beginning of your
> dialplan to set the sip_cid_type accordingly.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
> Service Client IPeva
> tel: 0811 46 26 26
> *www.ipeva.fr* - *www.ipeva-studio.com*
>
> *Ce message et toutes les pièces jointes sont confidentiels et établis à
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autorisée est interdite. Tout message électronique est susceptible
> d'altération. IPeva décline toute responsabilité au titre de ce message
> s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de
> ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.*
> * *
>
>
>
> Le 25/05/2010 à 12:13, Frank @ Impact a écrit :
>
>
> I can only find documentation on setting this value at the bridge level.
>
> Is there a way to control this setting in the gateway profile instead of
> having to do it in the bridge setup?
>
>
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> --
> Anthony Minessale II
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--
Anthony Minessale II
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