Can I use a playback_delimiter file list for a conference play operation?
Hi, I would like to play a list of files into a conference in one command. Is this possible? Thanks, David
The conference play command doesn't use the same core functions as the regular playback app so you can't use !-separated lists. However, you could use loopback to call into the conference and use the playback app. Just make sure that you disable the enter and exit sounds. Off the top of my head I'd say it looks something like this: originate {conference_enter_sound=silence_stream://1,conference_exit_sound=silence_stream://1}loopback/3000 &playback(file1.wav!file2.wav!file3.wav) Please read the standard disclaimer<http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer>before telling that it doesn't work. :) -MC On Mon, Jul 15, 2013 at 3:47 PM, D D <tru083@yahoo.com> wrote:
Hi,
I would like to play a list of files into a conference in one command. Is this possible?
Thanks, David
_________________________________________________________________________ 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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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Dear all, I am trying to setup a Grandstream gateway .. I set it for inbound registration .. Example : It registers through extension 1005 Now I need to do the following: if i call 99+7 digits, the call should be sent to the gateway ... Call Routing/Dial Plan (1 Stage Dialing Only) and Dial Settings : {x+} x - any digit from 0-9; I don't know if the number to be dialed has to be sent into the SIP message to extension 1005 or how should it be sent? I searched over the internet and found the following dialing plan, however it needs the ip address of the gateway which i don't have <include> <extension name="ht503-fxo-local"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^99(d{7})$"> <action application="set" data="effective_caller_id_number=0321234567"/> <action application="set" data="effective_caller_id_name=ThisIsMy Company"/> <action application="set" data="ignore_early_media=ring_ready"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="sofia/gateway/ht503-fxo/$1@192.168.0.9:5062"/> </condition> </extension> </include> could you please help me to set it correctly? thank you kind regards, S.Omran
Hello Sherif, I do not think you need to use the IP address of the gateway if you have already defined the gateway? I am using Grandstream HT503 and my dialplan is: <extension name="local"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/${default_gateway}/0${default_areacode}$1"/> </condition> </extension> HT503 is defined as default gateway. Thanks, Raj On Wednesday 17 July 2013 12:29 AM, Sherif Omran wrote:
Dear all,
I am trying to setup a Grandstream gateway .. I set it for inbound registration .. Example : It registers through extension 1005
Now I need to do the following:
if i call 99+7 digits, the call should be sent to the gateway ...
Call Routing/Dial Plan (1 Stage Dialing Only) and Dial Settings : {x+} x - any digit from 0-9;
I don't know if the number to be dialed has to be sent into the SIP message to extension 1005 or how should it be sent?
I searched over the internet and found the following dialing plan, however it needs the ip address of the gateway which i don't have
<include> <extension name="ht503-fxo-local"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^99(d{7})$"> <action application="set" data="effective_caller_id_number=0321234567"/> <action application="set" data="effective_caller_id_name=ThisIsMy Company"/> <action application="set" data="ignore_early_media=ring_ready"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="sofia/gateway/ht503-fxo/$1@192.168.0.9:5062"/> </condition> </extension> </include>
could you please help me to set it correctly?
thank you
kind regards, S.Omran
_________________________________________________________________________ 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
Standard file_string:// should also work, try conference 3000 play file_string:///tmp/1.wav!/tmp/2.wav!/tmp/3.wav -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Wednesday, July 17, 2013 at 2:22 AM, Michael Collins wrote:
The conference play command doesn't use the same core functions as the regular playback app so you can't use !-separated lists. However, you could use loopback to call into the conference and use the playback app. Just make sure that you disable the enter and exit sounds. Off the top of my head I'd say it looks something like this:
originate {conference_enter_sound=silence_stream://1,conference_exit_sound=silence_stream://1}loopback/3000 &playback(file1.wav!file2.wav!file3.wav)
Please read the standard disclaimer (http://wiki.freeswitch.org/wiki/Reporting_Bugs#Standard_Disclaimer) before telling that it doesn't work. :)
-MC
On Mon, Jul 15, 2013 at 3:47 PM, D D <tru083@yahoo.com (mailto:tru083@yahoo.com)> wrote:
Hi,
I would like to play a list of files into a conference in one command. Is this possible?
Thanks, David
_________________________________________________________________________ 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
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ 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
participants (5)
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D D
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Michael Collins
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Raj Saini
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Seven Du
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Sherif Omran