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I can publish to your demo wowza server with no problem and it works fine - it is definitely in the authentication - your flash server has no authentication for stream name 'live' and the system works fine - also I dont know where the logs are for the traffic and the wait case -

We have been using this software for over 2 years - this is a new request from our client - I posted another question similiar to this - can we do this through zoom??? in other words have our client connect to zoom and then capture the zoom video / audio feed and bridge that to rtmp?

No go - tried a few iterations - rtmp://wowza01.xyvid.com:1935/appname?uname:pass - this one works when we use FMLE to stream - but not in the program rtmp://username:password@rtmp.myserver.com:1935/appname - this is a no go - here is part of the server log when we use...

Our wowza servers are secure and require a username and password to push an rtmp stream. There is an accepted format that wowza accepts and it is - rtmp://rtmp.myserver.com:1935/application?username&password this does not work with the sample application when simply entering it into the server...

I got the system to work last night but I was under the impression that we could connect to a zoom session via a sip@domain.com address ? Is this not possible? Ultimately we are trying to connect telepresence devices to an RTMP wowza streaming solution. They use h264 via a sip audio/video...

We just sent our logs to you - we are trying to use SIP addresses as well as phone numbers to create an RTMP bridge - thanks for the help -

Cant get SIP working - - system says Established but won't dial a number to a cellphone - using version 5.2 on Centos - all ports open - what log files do you need to see - and is there a testing SIP provider that you recommend?

I have a client that has a bunch of DX80 cisco video conferencing units - can the flashphoner server connect to them and then restream the video and audio out to an RTMP Wowza Server ? I know the RTMP side is OK it is the SIP video calls that I am unsure of. The cisco unit is h.323 and SIP...

I sent an email to sales detailing what we want to do - thanks for the help you guys are really responsive...

We have installed your SIP to RTMP Flashphoner application and have successfully implemented it into our application. However there is one feature we need and were wondering how you would handle the situation. We need to place the call on hold and not broadcast out the rtmp encoding while the...

nevermind I found it -

How do I edit flashphoner.properties file - my demo server still does not work - do I need to do this on the server?

So I have sent the dumps and performed all of the recommendations - can you log onto the EC2 instance for ma and troubleshoot? I can give the current IP - I have tried all of the recommendations - except for the last one which told me to update to the latest version using service webcallserver...

Im hosting on EC2 and I don't really have access to download files from thier servers - I did manage to get the SIP part working but I am having trouble streaming to our wowza server - here is my command line - will this work? rtmp://UNAME:PASSWD@wowza01.xyvid.com/XYVI001 STREAM1 While...

My SIP line is fine but I keep getting an error (I think) that looks like this uASQOF-Qx2LA03-ijokpgCS-JhCPEU >>> XYVI001:XYVI001@rtmp://wowza01.xyvid.com:1935/XYVI001 FINISHED Any Ideas?