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?
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?
_______________________________________________ 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
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?
_______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Anthoney, Maybe you replied to the wrong thread or I am missing the connection between sip_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 to sip_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] 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? _______________________________________________ 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 _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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 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 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 ] 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?
_______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _______________________________________________ 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
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
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 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?
_______________________________________________ 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
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 _______________________________________________ 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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
I tried putting <param name="caller-id-type" value="pid"/> in the profile settings section and also in the profile gateway section. And in each case, FS still use rpid and seemed to ignore the param setting. I tested this on a fresh git from today. Am I putting it in the wrong place? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, May 27, 2010 9:45 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] sip_cid_type settings 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"/>
it will prefer the type used on the inbound leg. unless you override it with the var. On Fri, May 28, 2010 at 9:27 AM, Frank @ Impact <frank@impactfax.com> wrote:
I tried putting
<param name="caller-id-type" value="pid"/>
in the profile settings section and also in the profile gateway section. And in each case, FS still use rpid and seemed to ignore the param setting.
I tested this on a fresh git from today. Am I putting it in the wrong place?
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, May 27, 2010 9:45 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] sip_cid_type settings
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"/>
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Is the only way to override it by setting the var for the bridge? "sip_cid_type=pid" ? is there no way to override the inbound leg with the profile/gateway settings? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, May 28, 2010 11:10 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] sip_cid_type settings it will prefer the type used on the inbound leg. unless you override it with the var. On Fri, May 28, 2010 at 9:27 AM, Frank @ Impact <frank@impactfax.com> wrote: I tried putting <param name="caller-id-type" value="pid"/> in the profile settings section and also in the profile gateway section. And in each case, FS still use rpid and seemed to ignore the param setting. I tested this on a fresh git from today. Am I putting it in the wrong place? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, May 27, 2010 9:45 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] sip_cid_type settings 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"/> _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
you can set the global var so it's always set on every call in vars.xml On Fri, May 28, 2010 at 10:29 AM, Frank @ Impact <frank@impactfax.com>wrote:
Is the only way to override it by setting the var for the bridge?
"sip_cid_type=pid” ?
is there no way to override the inbound leg with the profile/gateway settings?
-----Original Message----- *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Friday, May 28, 2010 11:10 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] sip_cid_type settings
it will prefer the type used on the inbound leg.
unless you override it with the var.
On Fri, May 28, 2010 at 9:27 AM, Frank @ Impact <frank@impactfax.com> wrote:
I tried putting
<param name="caller-id-type" value="pid"/>
in the profile settings section and also in the profile gateway section. And in each case, FS still use rpid and seemed to ignore the param setting.
I tested this on a fresh git from today. Am I putting it in the wrong place?
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Thursday, May 27, 2010 9:45 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] sip_cid_type settings
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"/>
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
participants (3)
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Anthony Minessale
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David Ponzone
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Frank ＠ Impact