freeswitch + flashphoner web call server
Hi Guys, Have anyone tried using flashphoner web call server? I already have a freeswitch and a flashponer webcallserver running. My problem is how to make freeswitch use the flashphoner webcallserver. Which freeswitch xml file do I need to edit to make use of the flashphoner webcallserver. Im using a flex phone client for my flex desktop application. Do you have any suggestions on the best approach for sip to rtmp gateway? Regards, jdayola
Just curious to know if FreeSWITCH's mod_rtmp didn't fit your needs. I've tested some of Flashphoner's products against FreeSWITCH (the products sent SIP calls to my FreeSWITCH), but for my needs I never needed nor desired the extra hop in there. Of course, if you meant RTMFP instead of RTMP, then I could understand easily why you are using the extra hop. I don't remember exactly though how it was that I set that up as it was a couple years ago. My guess is that Flashphoner registered each client as a SIP registration to my FreeSWITCH. If that is the case, then you need to set up entries in the appropriate directory include directory under directory/${directoryname} -- Yossi Neiman On 02/23/2015 08:09 AM, jdayola@spingine.com wrote:
Hi Guys,
Have anyone tried using flashphoner web call server? I already have a freeswitch and a flashponer webcallserver running. My problem is how to make freeswitch use the flashphoner webcallserver. Which freeswitch xml file do I need to edit to make use of the flashphoner webcallserver. Im using a flex phone client for my flex desktop application. Do you have any suggestions on the best approach for sip to rtmp gateway?
Regards, jdayola
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Actually I was able to make calls using the flashphoner client that I registered to my freeswitch server. My problem is how to make it work using my own flex client. Im not entirely sure if I even need flashphoner web call server for my problem. Im using fusionpbx on top of freeswitch. I already had a working callcenter setup using xlite or zoiper for testing environment. When I started using a flex client as replacement for the xlite/zoiper client the agents can no longer receive the calls. I already enabled mod_rtmp but It looks like im missing something. I've already read the mod_rtmp enty but Im still a bit confused. Sorry Im still a noob at this. On 2015-02-24 12:32 am, I put the Who? in Mishehu wrote:
Just curious to know if FreeSWITCH's mod_rtmp didn't fit your needs. I've tested some of Flashphoner's products against FreeSWITCH (the products sent SIP calls to my FreeSWITCH), but for my needs I never needed nor desired the extra hop in there. Of course, if you meant RTMFP instead of RTMP, then I could understand easily why you are using the extra hop. I don't remember exactly though how it was that I set that up as it was a couple years ago. My guess is that Flashphoner registered each client as a SIP registration to my FreeSWITCH. If that is the case, then you need to set up entries in the appropriate directory include directory under directory/${directoryname}
-- Yossi Neiman
On 02/23/2015 08:09 AM, jdayola@spingine.com wrote:
Hi Guys,
Have anyone tried using flashphoner web call server? I already have a freeswitch and a flashponer webcallserver running. My problem is how to make freeswitch use the flashphoner webcallserver. Which freeswitch xml file do I need to edit to make use of the flashphoner webcallserver. Im using a flex phone client for my flex desktop application. Do you have any suggestions on the best approach for sip to rtmp gateway?
Regards, jdayola
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
maybe webrtc and mod_verto would be a good solution for you?
On Feb 24, 2015, at 1:39 AM, jdayola@spingine.com wrote:
Actually I was able to make calls using the flashphoner client that I registered to my freeswitch server. My problem is how to make it work using my own flex client. Im not entirely sure if I even need flashphoner web call server for my problem. Im using fusionpbx on top of freeswitch. I already had a working callcenter setup using xlite or zoiper for testing environment. When I started using a flex client as replacement for the xlite/zoiper client the agents can no longer receive the calls. I already enabled mod_rtmp but It looks like im missing something. I've already read the mod_rtmp enty but Im still a bit confused. Sorry Im still a noob at this.
On 2015-02-24 12:32 am, I put the Who? in Mishehu wrote:
Just curious to know if FreeSWITCH's mod_rtmp didn't fit your needs. I've tested some of Flashphoner's products against FreeSWITCH (the products sent SIP calls to my FreeSWITCH), but for my needs I never needed nor desired the extra hop in there. Of course, if you meant RTMFP instead of RTMP, then I could understand easily why you are using the extra hop. I don't remember exactly though how it was that I set that up as it was a couple years ago. My guess is that Flashphoner registered each client as a SIP registration to my FreeSWITCH. If that is the case, then you need to set up entries in the appropriate directory include directory under directory/${directoryname}
-- Yossi Neiman
On 02/23/2015 08:09 AM, jdayola@spingine.com wrote:
Hi Guys,
Have anyone tried using flashphoner web call server? I already have a freeswitch and a flashponer webcallserver running. My problem is how to make freeswitch use the flashphoner webcallserver. Which freeswitch xml file do I need to edit to make use of the flashphoner webcallserver. Im using a flex phone client for my flex desktop application. Do you have any suggestions on the best approach for sip to rtmp gateway?
Regards, jdayola
WebRTC is now supported by many browsers natively or via plugins so it should work and is the future. If you still want legacy flash support I think you should read the mod_rtmp related js and/or flash code to find out how it works. I had made at least rtmplite and flash-videoio works with FS, and I had made a flowplayer plugin which even I made it support video. -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow <http://www.sparrowmailapp.com/?sig> On Wednesday, February 25, 2015 at 12:58 AM, Michael Jerris wrote: maybe webrtc and mod_verto would be a good solution for you? On Feb 24, 2015, at 1:39 AM, jdayola@spingine.com wrote: Actually I was able to make calls using the flashphoner client that I registered to my freeswitch server. My problem is how to make it work using my own flex client. Im not entirely sure if I even need flashphoner web call server for my problem. Im using fusionpbx on top of freeswitch. I already had a working callcenter setup using xlite or zoiper for testing environment. When I started using a flex client as replacement for the xlite/zoiper client the agents can no longer receive the calls. I already enabled mod_rtmp but It looks like im missing something. I've already read the mod_rtmp enty but Im still a bit confused. Sorry Im still a noob at this. On 2015-02-24 12:32 am, I put the Who? in Mishehu wrote: Just curious to know if FreeSWITCH's mod_rtmp didn't fit your needs. I've tested some of Flashphoner's products against FreeSWITCH (the products sent SIP calls to my FreeSWITCH), but for my needs I never needed nor desired the extra hop in there. Of course, if you meant RTMFP instead of RTMP, then I could understand easily why you are using the extra hop. I don't remember exactly though how it was that I set that up as it was a couple years ago. My guess is that Flashphoner registered each client as a SIP registration to my FreeSWITCH. If that is the case, then you need to set up entries in the appropriate directory include directory under directory/${directoryname} -- Yossi Neiman On 02/23/2015 08:09 AM, jdayola@spingine.com wrote: Hi Guys, Have anyone tried using flashphoner web call server? I already have a freeswitch and a flashponer webcallserver running. My problem is how to make freeswitch use the flashphoner webcallserver. Which freeswitch xml file do I need to edit to make use of the flashphoner webcallserver. Im using a flex phone client for my flex desktop application. Do you have any suggestions on the best approach for sip to rtmp gateway? Regards, jdayola _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 (4)
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I put the Who? in Mishehu
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jdayola＠spingine.com
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Michael Jerris
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Seven Du