[Freeswitch-users] Equivalent of "s" extension in Asterisk

Sun Jan 29 08:32:01 MSK 2012

It does not matter if the FXS port is a card in the FS server or an
analog gateway/ATA. The device to which the analog phone is connected
must provide the first sound that is heard (dial tone), or else connect
immediately via FS to the source of the desired sound file.

Some people call this "hot line", old telephone heads in the U.S. call
this "ringdown private line", Cisco turns this name around and calls it
PLAR (private line automatic ringdown) in their gateway devices.

With no special processing or commands you will hear the happy sound of
the dial tone, as always. I agree with the answer that says to call
automatically an IVR application that plays the desired file(s) and
processes the dialed digits in response.

Bote

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Saturday, 28 January, 2012 11:27
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk

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 at 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 at 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
<http://fonality.com/trixbox/forums/trixbox-forums/trixbox-endpoints/tri
xbox-batphone-mode>  by some people because it is seen in a film about
Batman J

Regards,

Brian

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

À : freeswitch-users at 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>
http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf

Best regards,

Brian

De : freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at 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 at 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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de
Saugort Dario Garcia Tovar
Envoyé : vendredi 27 janvier 2012 13:42
À : freeswitch-users at 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-td547
9581.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

-- 
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Consultor. 

CCCT, Nivel C2, Sector Yarey, Mz, 
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Caracas-Venezuela. 
Teléfono: +58 212 9081842 <tel:%2B58%20212%209081842>  
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________________________________________________________________________
_
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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________________________________________________________________________
_
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
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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 <tel:%2B58%20212%209081842>  
Cel: +58 412 2221515 <tel:%2B58%20412%202221515>  
dgarcia at anew.com.ve 
http://www.anew.com.ve

________________________________________________________________________
_
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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________________________________________________________________________
_
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http://www.freeswitchsolutions.com

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