Equivalent of "s" extension in Asterisk
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
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-... http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581... 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
_________________________________________________________________________ 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 <http://www.avg.com> Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12
-- 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
Hi Saugort, Thank you for these links. However that is not really what Im 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, Im quite new to FreeSwitch and Im trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I havent 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 _________________________________________________________________________ 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: 2109/4766 - Release Date: 01/25/12 -- 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
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-...
http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581...
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****
****
****
_________________________________________________________________________****
Professional FreeSWITCH Consulting Services:****
consulting@freeswitch.org****
http://www.freeswitchsolutions.com****
** **
FreeSWITCH-powered IP PBX: The CudaTel Communication Server****
** **
Official FreeSWITCH Sites****
http://wiki.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****
****
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, *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
Hi Avi, Maybe thats the term I should use, hot-dial. Im talking about analogic phones, not VoIP phones. If its 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 cards driver. In asterisks 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@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 Im 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, Im quite new to FreeSwitch and Im trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I havent 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 _________________________________________________________________________ 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: 2109/4766 - Release Date: 01/25/12 -- 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@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
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 <mailto: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> [mailto: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 <mailto: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-...
http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581...
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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 <http://www.avg.com> Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12
-- 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@anew.com.ve <mailto:dgarcia@anew.com.ve> http://www.anew.com.ve
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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
No virus found in this message. Checked by AVG - www.avg.com <http://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
Hi Saugort, Im 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/trixbox -batphone-mode> 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 thats the term I should use, hot-dial. Im talking about analogic phones, not VoIP phones. If its 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 cards driver. In asterisks 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@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 Im 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, Im quite new to FreeSwitch and Im trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I havent 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 _________________________________________________________________________ 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: 2109/4766 - Release Date: 01/25/12 -- 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@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 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
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<http://fonality.com/trixbox/forums/trixbox-forums/trixbox-endpoints/trixbox-batphone-mode>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<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-...
http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581...
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****
****
****
_________________________________________________________________________****
Professional FreeSWITCH Consulting Services:****
consulting@freeswitch.org****
http://www.freeswitchsolutions.com****
****
FreeSWITCH-powered IP PBX: The CudaTel Communication Server****
****
Official FreeSWITCH Sites****
http://wiki.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****
****
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, *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****
** **
Official FreeSWITCH Sites****
http://wiki.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****
****
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
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<http://fonality.com/trixbox/forums/trixbox-forums/trixbox-endpoints/trixbox-batphone-mode>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<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-...
http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581...
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****
****
****
_________________________________________________________________________****
Professional FreeSWITCH Consulting Services:****
consulting@freeswitch.org****
http://www.freeswitchsolutions.com****
****
FreeSWITCH-powered IP PBX: The CudaTel Communication Server****
****
Official FreeSWITCH Sites****
http://wiki.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****
****
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, *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****
** **
Official FreeSWITCH Sites****
http://wiki.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****
****
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
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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 <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@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> 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-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 ________________________________________________________________________ _ 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: 2109/4766 - Release Date: 01/25/12 -- 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@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 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@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
in your ftdm config for analog_spans, set the "hotline" param to the extension you want it to call. I don't think I see this anywhere on the wiki, would you mind adding it? Mike On Jan 29, 2012, at 12:32 AM, Bote Man wrote:
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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
Hi Avi, I'm testing on a Digium TDM400P card plugged directly to FreeSwitch Server. I know that this feature is supported by the driver because it works with Asterisk. Brian Le 28 janvier 2012 à 17:27, Avi Marcus <avi@avimarcus.net> a écrit :
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<http://fonality.com/trixbox/forums/trixbox-forums/trixbox-endpoints/trixbox-batphone-mode>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<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-...
http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581...
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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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****
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Professional FreeSWITCH Consulting Services:****
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server****
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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****
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IIRC Asterisk uses the "s" extension when a call enters a context without any specific destination information (i.e. a phone on an FXS port going off-hook). I'm not sure about the OpenZAP/Sangoma/etc stuff in FreeSWITCH but the dialplan equivalent of Asterisk "s" would be an extension with no condition: <extension name="playback"> <condition> <action application="whatever" data="example"/> </condition> </extension> On Thu, Jan 26, 2012 at 5:15 PM, Brian <brian@neotiq.com> 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
_________________________________________________________________________ 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
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-- Kristian Kielhofner
participants (8)
-
Avi Marcus
-
Bote Man
-
Brian
-
brian＠neotiq.com
-
Kristian Kielhofner
-
Michael Jerris
-
Ognjen Seslija
-
Saugort Dario Garcia Tovar