Brian, your phones are plugged directly into an FXS card in your FreeSWITCH server? What kind of card is it? This is something that the specific hardware/driver has to support, knowing which one you are using will be essential to giving you an answer..

-Avi


On Fri, Jan 27, 2012 at 11:16 PM, Ognjen Seslija <oseslija@gmail.com> wrote:
You can do this easy in Linksys/Cisco phones/ATAs with dialplan manipulation (i.e. (P0<:numbertocall>)). A number to call is a FS/Asterisk ext which does the IVR or whatever.


On Fri, Jan 27, 2012 at 6:05 PM, Brian <brian@neotiq.com> wrote:

Hi Saugort,

 

I’m trying to do the first one. When someone take the handset off-hook, before he has pressed any key, play some messages to him. That can be used to play some warning if the outgoing SIP trunk is down, for example. That feature is called batPhone by some people because it is seen in a film about Batman J

 

Regards,

Brian

 

De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Saugort Dario Garcia Tovar
Envoyé : vendredi 27 janvier 2012 17:23


À : freeswitch-users@lists.freeswitch.org
Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk

 

Hi Brian,

Could you explain a little bit more what your want do do?

Are you trying to do this:
the analogic phone is a FS extension, connected to an ATA device (or it is connected to FXS card in FS server?). When someone take the handset, automatically connect to another extension, make pstn call or play a message?

Or your are trying to do this:
An incoming call from a PSTN line connected to a TDM card in your FS server, you want to automatically connect to another extension, make pstn call or play a message?

On 1/27/2012 10:49 AM, Brian wrote:

Hi Avi,

 

Maybe that’s the term I should use, hot-dial. I’m talking about analogic phones, not VoIP phones.  If it’s an analogic phone plugged into an FXS socket, when you pick up the phone, the PBX is capable of detecting the off-hook event. That feature must be supported by the analogic card’s driver. In asterisk’s case, that is the option named “immediate=true|false” in the Zapata.conf file, as stated here: http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf

 

Best regards,

Brian

 

De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Avi Marcus
Envoyé : vendredi 27 janvier 2012 15:42
À : FreeSWITCH Users Help
Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk

 

I think you want a hot-dial..? Some phones/ATAs support it. I don't think it's an FS/Asterisk feature.

e.g. most phones don't even contact the sip server until the person "finished dialing".


-Avi




On Fri, Jan 27, 2012 at 4:38 PM, Brian <brian@neotiq.com> wrote:

Hi Saugort,

 

Thank you for these links. However that is not really what I’m looking for. What I am trying to do is to play a sound file to an analogic phone as soon as the later goes off-hook. What I called “eary media” is not really the ringtone as we often see in SIP protocol. Sorry for the confusion.

 

Any help would be greatly appreciated,

Regards,

Brian

 

De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Saugort Dario Garcia Tovar
Envoyé : vendredi 27 janvier 2012 13:42
À : freeswitch-users@lists.freeswitch.org
Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk

 

Hi Brian,

Take a look to this links:

http://wiki.freeswitch.org/wiki/Early_Media

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready

http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as-early-media-td2460205.html

http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581.html



On 1/26/2012 5:45 PM, Brian wrote:

Hi List,

 

I’m quite new to FreeSwitch and I’m trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I haven’t managed to figure it out how to do it with FreeSwitch. That is about the “s” extension in Asterisk dial plan. In the old Asterisk installation, when I pick up an analogic phone (plugged into an FXS slot of a Digium card)  the “s” extension gets run and I can play early media to the phone. Does anyone know if it is possible to achieve the same thing with FreeSwitch/FreeTDM?

 

Best regards,

 

Brian

 




_________________________________________________________________________
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No virus found in this message.
Checked by AVG - www.avg.com
Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12

 

--
Atentamente,
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Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Teléfono: +58 212 9081842
Cel: +58 412 2221515
dgarcia@anew.com.ve
http://www.anew.com.ve


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
 
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
 
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




No virus found in this message.
Checked by AVG - www.avg.com
Version: 2012.0.1901 / Virus Database: 2109/4768 - Release Date: 01/26/12

 

--
Atentamente,
Dario García
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Teléfono: +58 212 9081842
Cel: +58 412 2221515
dgarcia@anew.com.ve
http://www.anew.com.ve


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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