I first posted this query to, viewtopic.php?f=1&t=84926 , but didn’t get any response.
I believe I am encountering a similar problem as described in the above post. My situation is:
I have registered two Android phones as a SIP clients on AsteriskWin32 PBX using CSipSimple v.1.01.00r2272. I can dial from one extension to the other, but there is no audio when a call is made.
I am able to hear the audio of the PBX echo test, but the audio that is replayed is very poor, warbled. The handset mic and speaker media graphs both register variations during the test. I have also tried different settings on CSipSimple, but with no improvement.
I was able to use the above configuration successfully with Ozeki Phone System.
The device / system details are:
Handset 1: Micromax A90, Andriod 4.0.3
Handset 2: Samsung Galaxy GT-S5360, Andriod 2.3.6
AsteriskWin32 PBX running on Windows XP, SP3, .NET4.0
Please forgive my ignorance, I do not know much about altering the Asteriks source code. If the above explained fix is the solution, where do I find these files and what part of the code in the files (e.g. line number), do I edit?
Any information would be appreciated.