---------- Forwarded message ---------- From: sami sakly <it.samisakly@gmail.com> Date: 2014-08-13 22:56 GMT+02:00 Subject: Freeswitch Dialer using SIPml5 To: freeswitch-dev@lists.freeswitch.org As it appears from the thread title I'm new to Freeswitch and Voip solution. So please excuse my ignorance if I didn't get you from the first time. I'm building a Dialer as solution for small compaign based on freeswitch. The main idea is that when agent are present : I'm not sure they call fs server or it calls them and starting dialing customers from a database and whenever a client pick up the phone : it's automatically bridged to one of the free agent (already parked and waiting). What I have now ? : *Freeswitch & webrtc2sip working on my server. *SIPml5 Client issuing calls and working I tried also to write some php script to run from a client host , that calls 2 sip client and do the bridging. What I want to do now ? : I'm looking for a way to apply my business model automatically : 1- Calling agent and parking 2- Dialling customers and whenever it's picked up bridge to a free agent My questions : will it be possible to do this with only dialplans or I need to have some external script ? Where exactly I can use fs events ? I will need them on the php app to create control on dialling client parameters and for bridging. That seems a lot but I hope I will not be disappointed =) Thank you,
Taje a look at ICTDialer. It's free. Guillermo On Thursday, August 14, 2014, sami sakly <it.samisakly@gmail.com> wrote:
---------- Forwarded message ---------- From: sami sakly <it.samisakly@gmail.com <javascript:_e(%7B%7D,'cvml','it.samisakly@gmail.com');>> Date: 2014-08-13 22:56 GMT+02:00 Subject: Freeswitch Dialer using SIPml5 To: freeswitch-dev@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-dev@lists.freeswitch.org');>
As it appears from the thread title I'm new to Freeswitch and Voip solution. So please excuse my ignorance if I didn't get you from the first time. I'm building a Dialer as solution for small compaign based on freeswitch. The main idea is that when agent are present : I'm not sure they call fs server or it calls them and starting dialing customers from a database and whenever a client pick up the phone : it's automatically bridged to one of the free agent (already parked and waiting).
What I have now ? :
*Freeswitch & webrtc2sip working on my server.
*SIPml5 Client issuing calls and working
I tried also to write some php script to run from a client host , that calls 2 sip client and do the bridging.
What I want to do now ? : I'm looking for a way to apply my business model automatically :
1- Calling agent and parking
2- Dialling customers and whenever it's picked up bridge to a free agent
My questions :
will it be possible to do this with only dialplans or I need to have some external script ?
Where exactly I can use fs events ? I will need them on the php app to create control on dialling client parameters and for bridging.
That seems a lot but I hope I will not be disappointed =)
Thank you,
-- Guillermo Ruiz Camauer
why use webrtc2sip? FreeSWITCH supports webrtc natively via either mod_verto and its associated js client, or via mod_sofia using something like sip.js (prefered) or sipml5 or jssip on the client side. Calling the agent and parking the call can be handled with something like mod_fifo or mod_callcenter (both offer similar auto call distribution functionality, but work differently) Dialing customers on the other hand would most likely require either an external program (script) connected to the ESL, or you would have to roll up your sleeves and dive into some C to write a call generator that links to your DB... While it is possible to do the external controls like this in PHP, that¹s probably not the most optimal language for doing so You could also checkout Newfies Dialer which is a supported open source dialer On 8/14/14 3:28 AM, "sami sakly" <it.samisakly@gmail.com> wrote:
---------- Forwarded message ---------- From: sami sakly <it.samisakly@gmail.com> Date: 2014-08-13 22:56 GMT+02:00 Subject: Freeswitch Dialer using SIPml5 To: freeswitch-dev@lists.freeswitch.org
As it appears from the thread title I'm new to Freeswitch and Voip solution. So please excuse my ignorance if I didn't get you from the first time. I'm building a Dialer as solution for small compaign based on freeswitch. The main idea is that when agent are present : I'm not sure they call fs server or it calls them and starting dialing customers from a database and whenever a client pick up the phone : it's automatically bridged to one of the free agent (already parked and waiting).
What I have now ? :
*Freeswitch & webrtc2sip working on my server.
*SIPml5 Client issuing calls and working
I tried also to write some php script to run from a client host , that calls 2 sip client and do the bridging.
What I want to do now ? : I'm looking for a way to apply my business model automatically :
1- Calling agent and parking
2- Dialling customers and whenever it's picked up bridge to a free agent
My questions :
will it be possible to do this with only dialplans or I need to have some external script ?
Where exactly I can use fs events ? I will need them on the php app to create control on dialling client parameters and for bridging.
That seems a lot but I hope I will not be disappointed =)
Thank you,
_________________________________________________________________________ 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
-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH
Thanks for your reply all =). I installed webrtc2sip at the beginning of this "journey" since I did not know what I'm going to use on Client side. There's no way I can use some already dialer because creating a dialer is my main purpose. The company I'm working in, had already developed some thing similar few years ago but with asterisk on server side. Where I can find details or examples for mod_fifo or mod_callcenter in use ? I've succeed to get event from server via websocket (esl) but I have no idea which event I should look for to issue calls to the signed in agent. My 'event plain ALL' list for a SIPml5 login then log out looks like : ''CUSTOM CUSTOM CUSTOM MESSAGE_QUERY MESSAGE_WAITING HEARTBEATRE_SCHEDULE CUSTOM PRESENCE_INCUSTOM" Regards, Sami 2014-08-14 15:58 GMT+02:00 Ken Rice <krice@freeswitch.org>:
why use webrtc2sip? FreeSWITCH supports webrtc natively via either mod_verto and its associated js client, or via mod_sofia using something like sip.js (prefered) or sipml5 or jssip on the client side.
Calling the agent and parking the call can be handled with something like mod_fifo or mod_callcenter (both offer similar auto call distribution functionality, but work differently)
Dialing customers on the other hand would most likely require either an external program (script) connected to the ESL, or you would have to roll up your sleeves and dive into some C to write a call generator that links to your DB...
While it is possible to do the external controls like this in PHP, that’s probably not the most optimal language for doing so
You could also checkout Newfies Dialer which is a supported open source dialer
On 8/14/14 3:28 AM, "sami sakly" <it.samisakly@gmail.com> wrote:
---------- Forwarded message ---------- From: *sami sakly* <it.samisakly@gmail.com> Date: 2014-08-13 22:56 GMT+02:00 Subject: Freeswitch Dialer using SIPml5 To: freeswitch-dev@lists.freeswitch.org
As it appears from the thread title I'm new to Freeswitch and Voip solution. So please excuse my ignorance if I didn't get you from the first time. I'm building a Dialer as solution for small compaign based on freeswitch. The main idea is that when agent are present : I'm not sure they call fs server or it calls them and starting dialing customers from a database and whenever a client pick up the phone : it's automatically bridged to one of the free agent (already parked and waiting).
What I have now ? :
*Freeswitch & webrtc2sip working on my server.
*SIPml5 Client issuing calls and working
I tried also to write some php script to run from a client host , that calls 2 sip client and do the bridging.
What I want to do now ? : I'm looking for a way to apply my business model automatically :
1- Calling agent and parking
2- Dialling customers and whenever it's picked up bridge to a free agent
My questions :
will it be possible to do this with only dialplans or I need to have some external script ?
Where exactly I can use fs events ? I will need them on the php app to create control on dialling client parameters and for bridging.
That seems a lot but I hope I will not be disappointed =)
Thank you,
------------------------------ _________________________________________________________________________ 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
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
_________________________________________________________________________ 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
If you don't know how to build a dialer and a pre-built open-source product is available, why head down the road of a doomed custom development project? On 14 August 2014 11:28, sami sakly <it.samisakly@gmail.com> wrote:
Thanks for your reply all =).
I installed webrtc2sip at the beginning of this "journey" since I did not know what I'm going to use on Client side. There's no way I can use some already dialer because creating a dialer is my main purpose. The company I'm working in, had already developed some thing similar few years ago but with asterisk on server side.
Where I can find details or examples for mod_fifo or mod_callcenter in use ?
I've succeed to get event from server via websocket (esl) but I have no idea which event I should look for to issue calls to the signed in agent. My 'event plain ALL' list for a SIPml5 login then log out looks like :
''CUSTOM CUSTOM CUSTOM MESSAGE_QUERY MESSAGE_WAITING HEARTBEATRE_SCHEDULE CUSTOM PRESENCE_INCUSTOM"
Regards, Sami
2014-08-14 15:58 GMT+02:00 Ken Rice <krice@freeswitch.org>:
why use webrtc2sip? FreeSWITCH supports webrtc natively via either
mod_verto and its associated js client, or via mod_sofia using something like sip.js (prefered) or sipml5 or jssip on the client side.
Calling the agent and parking the call can be handled with something like mod_fifo or mod_callcenter (both offer similar auto call distribution functionality, but work differently)
Dialing customers on the other hand would most likely require either an external program (script) connected to the ESL, or you would have to roll up your sleeves and dive into some C to write a call generator that links to your DB...
While it is possible to do the external controls like this in PHP, that’s probably not the most optimal language for doing so
You could also checkout Newfies Dialer which is a supported open source dialer
On 8/14/14 3:28 AM, "sami sakly" <it.samisakly@gmail.com> wrote:
---------- Forwarded message ---------- From: *sami sakly* <it.samisakly@gmail.com> Date: 2014-08-13 22:56 GMT+02:00 Subject: Freeswitch Dialer using SIPml5 To: freeswitch-dev@lists.freeswitch.org
As it appears from the thread title I'm new to Freeswitch and Voip solution. So please excuse my ignorance if I didn't get you from the first time. I'm building a Dialer as solution for small compaign based on freeswitch. The main idea is that when agent are present : I'm not sure they call fs server or it calls them and starting dialing customers from a database and whenever a client pick up the phone : it's automatically bridged to one of the free agent (already parked and waiting).
What I have now ? :
*Freeswitch & webrtc2sip working on my server.
*SIPml5 Client issuing calls and working
I tried also to write some php script to run from a client host , that calls 2 sip client and do the bridging.
What I want to do now ? : I'm looking for a way to apply my business model automatically :
1- Calling agent and parking
2- Dialling customers and whenever it's picked up bridge to a free agent
My questions :
will it be possible to do this with only dialplans or I need to have some external script ?
Where exactly I can use fs events ? I will need them on the php app to create control on dialling client parameters and for bridging.
That seems a lot but I hope I will not be disappointed =)
Thank you,
------------------------------ _________________________________________________________________________ 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
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
_________________________________________________________________________ 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
I'm with Ben on this one. Building a dialer "from scratch" as it were is a huge task. I think learning an existing dialer like ICTDialer or Newfies would be a much better use of your time. Is there perhaps a very compelling reason to build your own custom dialer? If so it needs to be VERY compelling. -MC On Thu, Aug 14, 2014 at 8:43 AM, Ben Langfeld <ben@langfeld.co.uk> wrote:
If you don't know how to build a dialer and a pre-built open-source product is available, why head down the road of a doomed custom development project?
On 14 August 2014 11:28, sami sakly <it.samisakly@gmail.com> wrote:
Thanks for your reply all =).
I installed webrtc2sip at the beginning of this "journey" since I did not know what I'm going to use on Client side. There's no way I can use some already dialer because creating a dialer is my main purpose. The company I'm working in, had already developed some thing similar few years ago but with asterisk on server side.
Where I can find details or examples for mod_fifo or mod_callcenter in use ?
I've succeed to get event from server via websocket (esl) but I have no idea which event I should look for to issue calls to the signed in agent. My 'event plain ALL' list for a SIPml5 login then log out looks like :
''CUSTOM CUSTOM CUSTOM MESSAGE_QUERY MESSAGE_WAITING HEARTBEATRE_SCHEDULE CUSTOM PRESENCE_INCUSTOM"
Regards, Sami
2014-08-14 15:58 GMT+02:00 Ken Rice <krice@freeswitch.org>:
why use webrtc2sip? FreeSWITCH supports webrtc natively via either
mod_verto and its associated js client, or via mod_sofia using something like sip.js (prefered) or sipml5 or jssip on the client side.
Calling the agent and parking the call can be handled with something like mod_fifo or mod_callcenter (both offer similar auto call distribution functionality, but work differently)
Dialing customers on the other hand would most likely require either an external program (script) connected to the ESL, or you would have to roll up your sleeves and dive into some C to write a call generator that links to your DB...
While it is possible to do the external controls like this in PHP, that’s probably not the most optimal language for doing so
You could also checkout Newfies Dialer which is a supported open source dialer
On 8/14/14 3:28 AM, "sami sakly" <it.samisakly@gmail.com> wrote:
---------- Forwarded message ---------- From: *sami sakly* <it.samisakly@gmail.com> Date: 2014-08-13 22:56 GMT+02:00 Subject: Freeswitch Dialer using SIPml5 To: freeswitch-dev@lists.freeswitch.org
As it appears from the thread title I'm new to Freeswitch and Voip solution. So please excuse my ignorance if I didn't get you from the first time. I'm building a Dialer as solution for small compaign based on freeswitch. The main idea is that when agent are present : I'm not sure they call fs server or it calls them and starting dialing customers from a database and whenever a client pick up the phone : it's automatically bridged to one of the free agent (already parked and waiting).
What I have now ? :
*Freeswitch & webrtc2sip working on my server.
*SIPml5 Client issuing calls and working
I tried also to write some php script to run from a client host , that calls 2 sip client and do the bridging.
What I want to do now ? : I'm looking for a way to apply my business model automatically :
1- Calling agent and parking
2- Dialling customers and whenever it's picked up bridge to a free agent
My questions :
will it be possible to do this with only dialplans or I need to have some external script ?
Where exactly I can use fs events ? I will need them on the php app to create control on dialling client parameters and for bridging.
That seems a lot but I hope I will not be disappointed =)
Thank you,
------------------------------ _________________________________________________________________________ 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
-- Ken
*http://www.FreeSWITCH.org <http://www.FreeSWITCH.org> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org <http://www.OSTAG.org> *irc.freenode.net #freeswitch Twitter: @FreeSWITCH
_________________________________________________________________________ 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
_________________________________________________________________________ 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
participants (5)
-
Ben Langfeld
-
Guillermo Ruiz Camauer
-
Ken Rice
-
Michael Collins
-
sami sakly