Hello: I've been using asterisk for a little bit over three years now. A couple of months ago I found out about freeswitch, took a look at it, thought it was interesting and moved on. A few weeks ago, I started looking at a project I've been wanting to work on for quite a while using conferences and started exploring systems and different approaches. Based on the requirements I have, I decided to use freeswitch. It seemed like it had the best support for conferencing so I went for it. According to some documentation I found it also seems to allow for more concurrent calls than asterisk which is an added bonus. I got a server ready, installed FC8 on it which is what I have in production now, unpacked freeswitch there and so far it's running beautifully. Very painless process really to get it installed, I was happy to see that. Configuration seems a bit different since it's XML; but being a developer myself I can see many advantages to having done that in the future as the system scales and grows in complexity. Sorry for the long introduction, getting to my question now. So ... What I want to be able to do is the following: Create and control conferences via the HTTP API. I've been reading a bit for the past two days the documentation and I am becoming more familiar now with how things are done using ESL, the support for PHP, perl and I believe others. a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference? b) When a conference is created, or when I go to create a new conference via HTTP using the API, does it allow for example for all numbers that will be added to be dialed at once? Or should the process be dial each participant, sending say 3 http requests via the API? The API command "conference dial" seems to only take one argument for destination number; but I am asking just in case I missed something. Thanks in advance for the help and I apologize for the long email. Alfredo
look at the "mad boss" extension in the default dialplan conf/dialplan/default.xml to see how to craft an all-hands conference. otherwise individual calls to originate to send people to the conference is also ok. On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com
wrote:
Hello:
I've been using asterisk for a little bit over three years now. A couple of months ago I found out about freeswitch, took a look at it, thought it was interesting and moved on. A few weeks ago, I started looking at a project I've been wanting to work on for quite a while using conferences and started exploring systems and different approaches. Based on the requirements I have, I decided to use freeswitch. It seemed like it had the best support for conferencing so I went for it. According to some documentation I found it also seems to allow for more concurrent calls than asterisk which is an added bonus.
I got a server ready, installed FC8 on it which is what I have in production now, unpacked freeswitch there and so far it's running beautifully. Very painless process really to get it installed, I was happy to see that. Configuration seems a bit different since it's XML; but being a developer myself I can see many advantages to having done that in the future as the system scales and grows in complexity.
Sorry for the long introduction, getting to my question now. So ... What I want to be able to do is the following:
Create and control conferences via the HTTP API. I've been reading a bit for the past two days the documentation and I am becoming more familiar now with how things are done using ESL, the support for PHP, perl and I believe others.
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
b) When a conference is created, or when I go to create a new conference via HTTP using the API, does it allow for example for all numbers that will be added to be dialed at once? Or should the process be dial each participant, sending say 3 http requests via the API? The API command "conference dial" seems to only take one argument for destination number; but I am asking just in case I missed something.
Thanks in advance for the help and I apologize for the long email.
Alfredo
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Appreciate the fast response Anthony. Response or ideas on how to implement a) ? a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference? Thanks in advance, Alfredo Q-V On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
look at the "mad boss" extension in the default dialplan conf/dialplan/default.xml to see how to craft an all-hands conference. otherwise individual calls to originate to send people to the conference is also ok.
On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Hello:
I've been using asterisk for a little bit over three years now. A couple of months ago I found out about freeswitch, took a look at it, thought it was interesting and moved on. A few weeks ago, I started looking at a project I've been wanting to work on for quite a while using conferences and started exploring systems and different approaches. Based on the requirements I have, I decided to use freeswitch. It seemed like it had the best support for conferencing so I went for it. According to some documentation I found it also seems to allow for more concurrent calls than asterisk which is an added bonus.
I got a server ready, installed FC8 on it which is what I have in production now, unpacked freeswitch there and so far it's running beautifully. Very painless process really to get it installed, I was happy to see that. Configuration seems a bit different since it's XML; but being a developer myself I can see many advantages to having done that in the future as the system scales and grows in complexity.
Sorry for the long introduction, getting to my question now. So ... What I want to be able to do is the following:
Create and control conferences via the HTTP API. I've been reading a bit for the past two days the documentation and I am becoming more familiar now with how things are done using ESL, the support for PHP, perl and I believe others.
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
b) When a conference is created, or when I go to create a new conference via HTTP using the API, does it allow for example for all numbers that will be added to be dialed at once? Or should the process be dial each participant, sending say 3 http requests via the API? The API command "conference dial" seems to only take one argument for destination number; but I am asking just in case I missed something.
Thanks in advance for the help and I apologize for the long email.
Alfredo
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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the flags are set as part of the dial string so you can easily choose that, int the example I told you to look at notice the +flags{} bit at the end of some of the dial strings. On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil < lawwton@gmail.com> wrote:
Appreciate the fast response Anthony.
Response or ideas on how to implement a) ?
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
Thanks in advance,
Alfredo Q-V
On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
look at the "mad boss" extension in the default dialplan conf/dialplan/default.xml to see how to craft an all-hands conference. otherwise individual calls to originate to send people to the conference is also ok.
On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Hello:
I've been using asterisk for a little bit over three years now. A couple of months ago I found out about freeswitch, took a look at it, thought it was interesting and moved on. A few weeks ago, I started looking at a project I've been wanting to work on for quite a while using conferences and started exploring systems and different approaches. Based on the requirements I have, I decided to use freeswitch. It seemed like it had the best support for conferencing so I went for it. According to some documentation I found it also seems to allow for more concurrent calls than asterisk which is an added bonus.
I got a server ready, installed FC8 on it which is what I have in production now, unpacked freeswitch there and so far it's running beautifully. Very painless process really to get it installed, I was happy to see that. Configuration seems a bit different since it's XML; but being a developer myself I can see many advantages to having done that in the future as the system scales and grows in complexity.
Sorry for the long introduction, getting to my question now. So ... What I want to be able to do is the following:
Create and control conferences via the HTTP API. I've been reading a bit for the past two days the documentation and I am becoming more familiar now with how things are done using ESL, the support for PHP, perl and I believe others.
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
b) When a conference is created, or when I go to create a new conference via HTTP using the API, does it allow for example for all numbers that will be added to be dialed at once? Or should the process be dial each participant, sending say 3 http requests via the API? The API command "conference dial" seems to only take one argument for destination number; but I am asking just in case I missed something.
Thanks in advance for the help and I apologize for the long email.
Alfredo
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http://lists.freeswitch.org/mailman/options/freeswitch-users
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Awesome! That's even nicer. Appreciate it. Alfredo Q-V On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
the flags are set as part of the dial string so you can easily choose that, int the example I told you to look at notice the +flags{} bit at the end of some of the dial strings.
On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Appreciate the fast response Anthony.
Response or ideas on how to implement a) ?
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
Thanks in advance,
Alfredo Q-V
On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
look at the "mad boss" extension in the default dialplan conf/dialplan/default.xml to see how to craft an all-hands conference. otherwise individual calls to originate to send people to the conference is also ok.
On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Hello:
I've been using asterisk for a little bit over three years now. A couple of months ago I found out about freeswitch, took a look at it, thought it was interesting and moved on. A few weeks ago, I started looking at a project I've been wanting to work on for quite a while using conferences and started exploring systems and different approaches. Based on the requirements I have, I decided to use freeswitch. It seemed like it had the best support for conferencing so I went for it. According to some documentation I found it also seems to allow for more concurrent calls than asterisk which is an added bonus.
I got a server ready, installed FC8 on it which is what I have in production now, unpacked freeswitch there and so far it's running beautifully. Very painless process really to get it installed, I was happy to see that. Configuration seems a bit different since it's XML; but being a developer myself I can see many advantages to having done that in the future as the system scales and grows in complexity.
Sorry for the long introduction, getting to my question now. So ... What I want to be able to do is the following:
Create and control conferences via the HTTP API. I've been reading a bit for the past two days the documentation and I am becoming more familiar now with how things are done using ESL, the support for PHP, perl and I believe others.
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
b) When a conference is created, or when I go to create a new conference via HTTP using the API, does it allow for example for all numbers that will be added to be dialed at once? Or should the process be dial each participant, sending say 3 http requests via the API? The API command "conference dial" seems to only take one argument for destination number; but I am asking just in case I missed something.
Thanks in advance for the help and I apologize for the long email.
Alfredo
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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Anthony: I think I spoke too soon. I looked at the example; but I was under the impression based on your previous comment that I would be able to invoke the conference dial api command over HTTP and specify the flags as part of the dial string. The wiki page has the following for the API call I am thinking: dial Dial a destination via a specific endpoint (ie. call mom from the conference). Usage: conference <confname> dial [{dial string options}]<endpoint_module_name>/<destination> [<callerid_number> [<callerid_name>]] I would like to specify the privilege in this api call. Is that doable? If not how could I accomplish it? Would I be able to pass the flags in {dial string options}? Thanks in advance for the help, Alfredo On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Awesome! That's even nicer.
Appreciate it.
Alfredo Q-V
On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
the flags are set as part of the dial string so you can easily choose that, int the example I told you to look at notice the +flags{} bit at the end of some of the dial strings.
On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Appreciate the fast response Anthony.
Response or ideas on how to implement a) ?
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
Thanks in advance,
Alfredo Q-V
On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
look at the "mad boss" extension in the default dialplan conf/dialplan/default.xml to see how to craft an all-hands conference. otherwise individual calls to originate to send people to the conference is also ok.
On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Hello:
I've been using asterisk for a little bit over three years now. A couple of months ago I found out about freeswitch, took a look at it, thought it was interesting and moved on. A few weeks ago, I started looking at a project I've been wanting to work on for quite a while using conferences and started exploring systems and different approaches. Based on the requirements I have, I decided to use freeswitch. It seemed like it had the best support for conferencing so I went for it. According to some documentation I found it also seems to allow for more concurrent calls than asterisk which is an added bonus.
I got a server ready, installed FC8 on it which is what I have in production now, unpacked freeswitch there and so far it's running beautifully. Very painless process really to get it installed, I was happy to see that. Configuration seems a bit different since it's XML; but being a developer myself I can see many advantages to having done that in the future as the system scales and grows in complexity.
Sorry for the long introduction, getting to my question now. So ... What I want to be able to do is the following:
Create and control conferences via the HTTP API. I've been reading a bit for the past two days the documentation and I am becoming more familiar now with how things are done using ESL, the support for PHP, perl and I believe others.
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
b) When a conference is created, or when I go to create a new conference via HTTP using the API, does it allow for example for all numbers that will be added to be dialed at once? Or should the process be dial each participant, sending say 3 http requests via the API? The API command "conference dial" seems to only take one argument for destination number; but I am asking just in case I missed something.
Thanks in advance for the help and I apologize for the long email.
Alfredo
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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look up the originate api instead of conference dial originate /dial/string conference:myconf+flags{foo} inline On Fri, Jan 15, 2010 at 1:18 PM, Alfredo Quiroga-Villamil <lawwton@gmail.com
wrote:
Anthony:
I think I spoke too soon. I looked at the example; but I was under the impression based on your previous comment that I would be able to invoke the conference dial api command over HTTP and specify the flags as part of the dial string. The wiki page has the following for the API call I am thinking:
dial
Dial a destination via a specific endpoint (ie. call mom from the conference).
Usage: conference <confname> dial [{dial string options}]<endpoint_module_name>/<destination> [<callerid_number> [<callerid_name>]]
I would like to specify the privilege in this api call. Is that doable? If not how could I accomplish it? Would I be able to pass the flags in {dial string options}?
Thanks in advance for the help,
Alfredo
On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Awesome! That's even nicer.
Appreciate it.
Alfredo Q-V
On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
the flags are set as part of the dial string so you can easily choose that, int the example I told you to look at notice the +flags{} bit at the end of some of the dial strings.
On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Appreciate the fast response Anthony.
Response or ideas on how to implement a) ?
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
Thanks in advance,
Alfredo Q-V
On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
look at the "mad boss" extension in the default dialplan conf/dialplan/default.xml to see how to craft an all-hands
conference.
otherwise individual calls to originate to send people to the conference is also ok.
On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Hello:
I've been using asterisk for a little bit over three years now. A couple of months ago I found out about freeswitch, took a look at
it,
thought it was interesting and moved on. A few weeks ago, I started looking at a project I've been wanting to work on for quite a while using conferences and started exploring systems and different approaches. Based on the requirements I have, I decided to use freeswitch. It seemed like it had the best support for conferencing so I went for it. According to some documentation I found it also seems to allow for more concurrent calls than asterisk which is an added bonus.
I got a server ready, installed FC8 on it which is what I have in production now, unpacked freeswitch there and so far it's running beautifully. Very painless process really to get it installed, I was happy to see that. Configuration seems a bit different since it's XML; but being a developer myself I can see many advantages to having done that in the future as the system scales and grows in complexity.
Sorry for the long introduction, getting to my question now. So ... What I want to be able to do is the following:
Create and control conferences via the HTTP API. I've been reading a bit for the past two days the documentation and I am becoming more familiar now with how things are done using ESL, the support for PHP, perl and I believe others.
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
b) When a conference is created, or when I go to create a new conference via HTTP using the API, does it allow for example for all numbers that will be added to be dialed at once? Or should the process be dial each participant, sending say 3 http requests via the API? The API command "conference dial" seems to only take one argument for destination number; but I am asking just in case I missed something.
Thanks in advance for the help and I apologize for the long email.
Alfredo
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Great! Thanks Anthony, really appreciate the help. Alfredo Q-V On Fri, Jan 15, 2010 at 3:28 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
look up the originate api instead of conference dial
originate /dial/string conference:myconf+flags{foo} inline
On Fri, Jan 15, 2010 at 1:18 PM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Anthony:
I think I spoke too soon. I looked at the example; but I was under the impression based on your previous comment that I would be able to invoke the conference dial api command over HTTP and specify the flags as part of the dial string. The wiki page has the following for the API call I am thinking:
dial
Dial a destination via a specific endpoint (ie. call mom from the conference).
Usage: conference <confname> dial [{dial string options}]<endpoint_module_name>/<destination> [<callerid_number> [<callerid_name>]]
I would like to specify the privilege in this api call. Is that doable? If not how could I accomplish it? Would I be able to pass the flags in {dial string options}?
Thanks in advance for the help,
Alfredo
On Fri, Jan 15, 2010 at 12:53 PM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Awesome! That's even nicer.
Appreciate it.
Alfredo Q-V
On Fri, Jan 15, 2010 at 12:02 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
the flags are set as part of the dial string so you can easily choose that, int the example I told you to look at notice the +flags{} bit at the end of some of the dial strings.
On Fri, Jan 15, 2010 at 10:42 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote:
Appreciate the fast response Anthony.
Response or ideas on how to implement a) ?
a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference?
Thanks in advance,
Alfredo Q-V
On Fri, Jan 15, 2010 at 11:13 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
look at the "mad boss" extension in the default dialplan conf/dialplan/default.xml to see how to craft an all-hands conference. otherwise individual calls to originate to send people to the conference is also ok.
On Fri, Jan 15, 2010 at 9:00 AM, Alfredo Quiroga-Villamil <lawwton@gmail.com> wrote: > > Hello: > > I've been using asterisk for a little bit over three years now. A > couple of months ago I found out about freeswitch, took a look at > it, > thought it was interesting and moved on. A few weeks ago, I started > looking at a project I've been wanting to work on for quite a while > using conferences and started exploring systems and different > approaches. Based on the requirements I have, I decided to use > freeswitch. It seemed like it had the best support for conferencing > so > I went for it. According to some documentation I found it also > seems > to allow for more concurrent calls than asterisk which is an added > bonus. > > I got a server ready, installed FC8 on it which is what I have in > production now, unpacked freeswitch there and so far it's running > beautifully. Very painless process really to get it installed, I > was > happy to see that. Configuration seems a bit different since it's > XML; > but being a developer myself I can see many advantages to having > done > that in the future as the system scales and grows in complexity. > > Sorry for the long introduction, getting to my question now. So ... > What I want to be able to do is the following: > > Create and control conferences via the HTTP API. I've been reading > a > bit for the past two days the documentation and I am becoming more > familiar now with how things are done using ESL, the support for > PHP, > perl and I believe others. > > a) It seemed to me like the way to setup the moderator of the > conference is by setting a parameter in the DialPlan and specifying > based on a condition who the moderator is, say for instance the > destination number. That's fine and it makes sense, however, say > that > I am creating a new conference and I want to have 3 participants > where > one of them is the moderator. What would I have to do to specify > that > person A dialing for example number xxx-xxx-xxxx is the moderator > (via > HTTP)? Would I have to create my own call to the system and add say > an > entry to DialPlan with the right parameter for the moderator, then > create the conference? > > b) When a conference is created, or when I go to create a new > conference via HTTP using the API, does it allow for example for > all > numbers that will be added to be dialed at once? Or should the > process > be dial each participant, sending say 3 http requests via the API? > The > API command "conference dial" seems to only take one argument for > destination number; but I am asking just in case I missed > something. > > Thanks in advance for the help and I apologize for the long email. > > Alfredo > > _______________________________________________ > 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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participants (2)
-
Alfredo Quiroga-Villamil
-
Anthony Minessale