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
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