Is there a way to enable paging?
We had paging setup when you dialed ext 350 on asterisk. Is there a way to enable any type of paging in free switch? Bill -- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
Paging in what respect? Being this vague about what you're asking will not really help us answer your questions. For example. The group page/multicast page on the SPA phones works with FreeSWITCH for sure. But outside of that i'm not too sure what you're wanting. /b On Jun 29, 2007, at 1:39 PM, Bill Locke wrote:
We had paging setup when you dialed ext 350 on asterisk. Is there a way to enable any type of paging in free switch?
Bill
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
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We lift the handset on our Grandstream GXP-2000 phone and dial 350 [send]. All other Grandstream phones then ring once and then go in to speaker/mute simultaneously, allowing us to address the entire building from a single phone. In asterisk, the method that allows this to happen is a conference room is created dynamically, all SIP endpoints are invited in to it with the parameter "Call-Info: answer-after=0", and another parameter which I'm not sure of which puts them on mute. A predefined macro in Asterisk addresses such problems as phones that are in use. The actual Asterisk application we are using is simply called Page(). Thanks Bill Brian West wrote:
Paging in what respect? Being this vague about what you're asking will not really help us answer your questions. For example. The group page/multicast page on the SPA phones works with FreeSWITCH for sure. But outside of that i'm not too sure what you're wanting.
/b
On Jun 29, 2007, at 1:39 PM, Bill Locke wrote:
We had paging setup when you dialed ext 350 on asterisk. Is there a way to enable any type of paging in free switch?
Bill
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
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-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
We have this <action application="set" data="sip_h_Call-Info: answer-after=0"/> added and working. The problem we are having now is getting all the phones to ring at 1 time. We are not sure how to do this portion now. If you dial 350 the first extension rings but no others. <!-- Dial 350 to page all SIP phones from x200-220 --> <extension name="PBX Extension"> <condition field="destination_number" expression="^350$"> <!-- This will dial a registered phone at the ip or domain name you set in the globals section --> <!-- (The % indicates it is an internal extension) --> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_h_Call-Info=answer-after=0"/> <action application="bridge" data="sofia/$${domain}/200%$${domain}"/> <action application="bridge" data="sofia/$${domain}/201%$${domain}"/> <action application="bridge" data="sofia/$${domain}/202%$${domain}"/> <action application="bridge" data="sofia/$${domain}/203%$${domain}"/> <action application="bridge" data="sofia/$${domain}/204%$${domain}"/> <action application="bridge" data="sofia/$${domain}/205%$${domain}"/> <action application="bridge" data="sofia/$${domain}/206%$${domain}"/> <action application="bridge" data="sofia/$${domain}/207%$${domain}"/> <action application="bridge" data="sofia/$${domain}/208%$${domain}"/> <action application="bridge" data="sofia/$${domain}/209%$${domain}"/> <action application="bridge" data="sofia/$${domain}/210%$${domain}"/> <action application="bridge" data="sofia/$${domain}/211%$${domain}"/> <action application="bridge" data="sofia/$${domain}/212%$${domain}"/> <action application="bridge" data="sofia/$${domain}/213%$${domain}"/> <action application="bridge" data="sofia/$${domain}/214%$${domain}"/> <action application="bridge" data="sofia/$${domain}/215%$${domain}"/> <action application="bridge" data="sofia/$${domain}/216%$${domain}"/> <action application="bridge" data="sofia/$${domain}/217%$${domain}"/> <action application="bridge" data="sofia/$${domain}/218%$${domain}"/> <action application="bridge" data="sofia/$${domain}/219%$${domain}"/> <action application="bridge" data="sofia/$${domain}/220%$${domain}"/> </condition> </extension> Brian West wrote:
Try this:
<action application="set" data="sip_h_Call-Info: answer-after=0"/>
Before your bridge command.
/b
On Jul 9, 2007, at 12:06 PM, Bill Locke wrote:
Call-Info: answer-after=0
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
Is your goal to have all phones offhook and listening to the announcement? /b On Jul 10, 2007, at 12:18 PM, Bill Locke wrote:
We have this <action application="set" data="sip_h_Call-Info: answer-after=0"/> added and working. The problem we are having now is getting all the phones to ring at 1 time. We are not sure how to do this portion now. If you dial 350 the first extension rings but no others.
<!-- Dial 350 to page all SIP phones from x200-220 --> <extension name="PBX Extension"> <condition field="destination_number" expression="^350$"> <!-- This will dial a registered phone at the ip or domain name you set in the globals section --> <!-- (The % indicates it is an internal extension) --> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_h_Call-Info=answer-after=0"/> <action application="bridge" data="sofia/$${domain}/200%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/201%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/202%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/203%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/204%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/205%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/206%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/207%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/208%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/209%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/210%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/211%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/212%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/213%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/214%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/215%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/216%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/217%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/218%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/219%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/220%$$ {domain}"/> </condition> </extension>
Brian West wrote:
Try this:
<action application="set" data="sip_h_Call-Info: answer-after=0"/>
Before your bridge command.
/b
On Jul 9, 2007, at 12:06 PM, Bill Locke wrote:
Call-Info: answer-after=0
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
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We want just phones that are not in use to kick into speaker phone and be muted at the same time. Bill Brian West wrote:
Is your goal to have all phones offhook and listening to the announcement?
/b
On Jul 10, 2007, at 12:18 PM, Bill Locke wrote:
We have this <action application="set" data="sip_h_Call-Info: answer-after=0"/> added and working. The problem we are having now is getting all the phones to ring at 1 time. We are not sure how to do this portion now. If you dial 350 the first extension rings but no others.
<!-- Dial 350 to page all SIP phones from x200-220 --> <extension name="PBX Extension"> <condition field="destination_number" expression="^350$"> <!-- This will dial a registered phone at the ip or domain name you set in the globals section --> <!-- (The % indicates it is an internal extension) --> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_h_Call-Info=answer-after=0"/> <action application="bridge" data="sofia/$${domain}/200%$${domain}"/> <action application="bridge" data="sofia/$${domain}/201%$${domain}"/> <action application="bridge" data="sofia/$${domain}/202%$${domain}"/> <action application="bridge" data="sofia/$${domain}/203%$${domain}"/> <action application="bridge" data="sofia/$${domain}/204%$${domain}"/> <action application="bridge" data="sofia/$${domain}/205%$${domain}"/> <action application="bridge" data="sofia/$${domain}/206%$${domain}"/> <action application="bridge" data="sofia/$${domain}/207%$${domain}"/> <action application="bridge" data="sofia/$${domain}/208%$${domain}"/> <action application="bridge" data="sofia/$${domain}/209%$${domain}"/> <action application="bridge" data="sofia/$${domain}/210%$${domain}"/> <action application="bridge" data="sofia/$${domain}/211%$${domain}"/> <action application="bridge" data="sofia/$${domain}/212%$${domain}"/> <action application="bridge" data="sofia/$${domain}/213%$${domain}"/> <action application="bridge" data="sofia/$${domain}/214%$${domain}"/> <action application="bridge" data="sofia/$${domain}/215%$${domain}"/> <action application="bridge" data="sofia/$${domain}/216%$${domain}"/> <action application="bridge" data="sofia/$${domain}/217%$${domain}"/> <action application="bridge" data="sofia/$${domain}/218%$${domain}"/> <action application="bridge" data="sofia/$${domain}/219%$${domain}"/> <action application="bridge" data="sofia/$${domain}/220%$${domain}"/> </condition> </extension>
Brian West wrote:
Try this:
<action application="set" data="sip_h_Call-Info: answer-after=0"/>
Before your bridge command.
/b
On Jul 9, 2007, at 12:06 PM, Bill Locke wrote:
Call-Info: answer-after=0
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
_______________________________________________ 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
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
Well you can't do a multi part page like that unless you originate them all into a conference first. /b On Jul 10, 2007, at 1:08 PM, Bill Locke wrote:
We want just phones that are not in use to kick into speaker phone and be muted at the same time.
Bill
Brian West wrote:
Is your goal to have all phones offhook and listening to the announcement?
/b
On Jul 10, 2007, at 12:18 PM, Bill Locke wrote:
We have this <action application="set" data="sip_h_Call-Info: answer-after=0"/> added and working. The problem we are having now is getting all the phones to ring at 1 time. We are not sure how to do this portion now. If you dial 350 the first extension rings but no others.
<!-- Dial 350 to page all SIP phones from x200-220 --> <extension name="PBX Extension"> <condition field="destination_number" expression="^350$"> <!-- This will dial a registered phone at the ip or domain name you set in the globals section --> <!-- (The % indicates it is an internal extension) --> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_h_Call-Info=answer- after=0"/> <action application="bridge" data="sofia/$${domain}/200%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/201%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/202%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/203%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/204%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/205%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/206%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/207%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/208%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/209%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/210%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/211%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/212%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/213%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/214%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/215%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/216%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/217%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/218%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/219%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/220%$$ {domain}"/> </condition> </extension>
Brian West wrote:
Try this:
<action application="set" data="sip_h_Call-Info: answer-after=0"/>
Before your bridge command.
/b
On Jul 9, 2007, at 12:06 PM, Bill Locke wrote:
Call-Info: answer-after=0
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
_______________________________________________ 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
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
This is how asterisk works. We are not sure how to do this with freeswitch. do you have an example? Bill Brian West wrote:
Well you can't do a multi part page like that unless you originate them all into a conference first.
/b
On Jul 10, 2007, at 1:08 PM, Bill Locke wrote:
We want just phones that are not in use to kick into speaker phone and be muted at the same time.
Bill
Brian West wrote:
Is your goal to have all phones offhook and listening to the announcement?
/b
On Jul 10, 2007, at 12:18 PM, Bill Locke wrote:
We have this <action application="set" data="sip_h_Call-Info: answer-after=0"/> added and working. The problem we are having now is getting all the phones to ring at 1 time. We are not sure how to do this portion now. If you dial 350 the first extension rings but no others.
<!-- Dial 350 to page all SIP phones from x200-220 --> <extension name="PBX Extension"> <condition field="destination_number" expression="^350$"> <!-- This will dial a registered phone at the ip or domain name you set in the globals section --> <!-- (The % indicates it is an internal extension) --> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_h_Call-Info=answer-after=0"/> <action application="bridge" data="sofia/$${domain}/200%$${domain}"/> <action application="bridge" data="sofia/$${domain}/201%$${domain}"/> <action application="bridge" data="sofia/$${domain}/202%$${domain}"/> <action application="bridge" data="sofia/$${domain}/203%$${domain}"/> <action application="bridge" data="sofia/$${domain}/204%$${domain}"/> <action application="bridge" data="sofia/$${domain}/205%$${domain}"/> <action application="bridge" data="sofia/$${domain}/206%$${domain}"/> <action application="bridge" data="sofia/$${domain}/207%$${domain}"/> <action application="bridge" data="sofia/$${domain}/208%$${domain}"/> <action application="bridge" data="sofia/$${domain}/209%$${domain}"/> <action application="bridge" data="sofia/$${domain}/210%$${domain}"/> <action application="bridge" data="sofia/$${domain}/211%$${domain}"/> <action application="bridge" data="sofia/$${domain}/212%$${domain}"/> <action application="bridge" data="sofia/$${domain}/213%$${domain}"/> <action application="bridge" data="sofia/$${domain}/214%$${domain}"/> <action application="bridge" data="sofia/$${domain}/215%$${domain}"/> <action application="bridge" data="sofia/$${domain}/216%$${domain}"/> <action application="bridge" data="sofia/$${domain}/217%$${domain}"/> <action application="bridge" data="sofia/$${domain}/218%$${domain}"/> <action application="bridge" data="sofia/$${domain}/219%$${domain}"/> <action application="bridge" data="sofia/$${domain}/220%$${domain}"/> </condition> </extension>
Brian West wrote:
Try this:
<action application="set" data="sip_h_Call-Info: answer-after=0"/>
Before your bridge command.
/b
On Jul 9, 2007, at 12:06 PM, Bill Locke wrote:
Call-Info: answer-after=0
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
_______________________________________________ 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
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
Do the phones support multicast paging in any way? Because we do multi cast sip invite on the SPA962's and talk to every phone on a multicast group. /b On Jul 10, 2007, at 1:28 PM, Bill Locke wrote:
This is how asterisk works. We are not sure how to do this with freeswitch. do you have an example?
Bill
Brian West wrote:
Well you can't do a multi part page like that unless you originate them all into a conference first.
/b
On Jul 10, 2007, at 1:08 PM, Bill Locke wrote:
We want just phones that are not in use to kick into speaker phone and be muted at the same time.
Bill
Brian West wrote:
Is your goal to have all phones offhook and listening to the announcement?
/b
On Jul 10, 2007, at 12:18 PM, Bill Locke wrote:
We have this <action application="set" data="sip_h_Call-Info: answer-after=0"/> added and working. The problem we are having now is getting all the phones to ring at 1 time. We are not sure how to do this portion now. If you dial 350 the first extension rings but no others.
<!-- Dial 350 to page all SIP phones from x200-220 --> <extension name="PBX Extension"> <condition field="destination_number" expression="^350$"> <!-- This will dial a registered phone at the ip or domain name you set in the globals section --> <!-- (The % indicates it is an internal extension) --> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_h_Call-Info=answer- after=0"/> <action application="bridge" data="sofia/$${domain}/200%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/201%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/202%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/203%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/204%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/205%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/206%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/207%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/208%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/209%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/210%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/211%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/212%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/213%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/214%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/215%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/216%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/217%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/218%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/219%$$ {domain}"/> <action application="bridge" data="sofia/$${domain}/220%$$ {domain}"/> </condition> </extension>
Brian West wrote:
Try this:
<action application="set" data="sip_h_Call-Info: answer- after=0"/>
Before your bridge command.
/b
On Jul 9, 2007, at 12:06 PM, Bill Locke wrote:
> Call-Info: answer-after=0
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
_______________________________________________ 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
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
I am not sure. We currently have asterisk setting up a dynamic conference room. it then initiates an invite to each phone on the list while passing the parameter to turn on speaker phone. does freeswitch not have anything similar? Bill Brian West wrote:
Do the phones support multicast paging in any way? Because we do multi cast sip invite on the SPA962's and talk to every phone on a multicast group.
/b
On Jul 10, 2007, at 1:28 PM, Bill Locke wrote:
This is how asterisk works. We are not sure how to do this with freeswitch. do you have an example?
Bill
Brian West wrote:
Well you can't do a multi part page like that unless you originate them all into a conference first.
/b
On Jul 10, 2007, at 1:08 PM, Bill Locke wrote:
We want just phones that are not in use to kick into speaker phone and be muted at the same time.
Bill
Brian West wrote:
Is your goal to have all phones offhook and listening to the announcement?
/b
On Jul 10, 2007, at 12:18 PM, Bill Locke wrote:
We have this <action application="set" data="sip_h_Call-Info: answer-after=0"/> added and working. The problem we are having now is getting all the phones to ring at 1 time. We are not sure how to do this portion now. If you dial 350 the first extension rings but no others.
<!-- Dial 350 to page all SIP phones from x200-220 --> <extension name="PBX Extension"> <condition field="destination_number" expression="^350$"> <!-- This will dial a registered phone at the ip or domain name you set in the globals section --> <!-- (The % indicates it is an internal extension) --> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_h_Call-Info=answer-after=0"/> <action application="bridge" data="sofia/$${domain}/200%$${domain}"/> <action application="bridge" data="sofia/$${domain}/201%$${domain}"/> <action application="bridge" data="sofia/$${domain}/202%$${domain}"/> <action application="bridge" data="sofia/$${domain}/203%$${domain}"/> <action application="bridge" data="sofia/$${domain}/204%$${domain}"/> <action application="bridge" data="sofia/$${domain}/205%$${domain}"/> <action application="bridge" data="sofia/$${domain}/206%$${domain}"/> <action application="bridge" data="sofia/$${domain}/207%$${domain}"/> <action application="bridge" data="sofia/$${domain}/208%$${domain}"/> <action application="bridge" data="sofia/$${domain}/209%$${domain}"/> <action application="bridge" data="sofia/$${domain}/210%$${domain}"/> <action application="bridge" data="sofia/$${domain}/211%$${domain}"/> <action application="bridge" data="sofia/$${domain}/212%$${domain}"/> <action application="bridge" data="sofia/$${domain}/213%$${domain}"/> <action application="bridge" data="sofia/$${domain}/214%$${domain}"/> <action application="bridge" data="sofia/$${domain}/215%$${domain}"/> <action application="bridge" data="sofia/$${domain}/216%$${domain}"/> <action application="bridge" data="sofia/$${domain}/217%$${domain}"/> <action application="bridge" data="sofia/$${domain}/218%$${domain}"/> <action application="bridge" data="sofia/$${domain}/219%$${domain}"/> <action application="bridge" data="sofia/$${domain}/220%$${domain}"/> </condition> </extension>
Brian West wrote: > Try this: > > <action application="set" data="sip_h_Call-Info: answer-after=0"/> > > Before your bridge command. > > /b > > > On Jul 9, 2007, at 12:06 PM, Bill Locke wrote: > >> Call-Info: answer-after=0 > > >
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
_______________________________________________ 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
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
-- Bill Locke AYALOGIC Systems administrator Personal Q 330-247-2182 Support Q 330-247-2188
bgapi originate {sip_h_Call-Info=answer-after=0}sofia/profile/ phone@ip &conference(1234) from event socket you can issue these... or from a javascript you can issue them via the execute if I recall. /b On Jul 10, 2007, at 2:12 PM, Bill Locke wrote:
I am not sure. We currently have asterisk setting up a dynamic conference room. it then initiates an invite to each phone on the list while passing the parameter to turn on speaker phone. does freeswitch not have anything similar?
Bill
Brian West wrote:
participants (2)
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