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@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@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
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*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 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@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org<freeswitch-users-bounces@lists.freeswitch.org> ] *On Behalf Of *Anthony Minessale *Sent:* Tuesday, May 25, 2010 11:01 AM *To:* freeswitch-users@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@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@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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