I have a arm machine with asterisk 1.8 and fpbx 2.10.
I have an IVR setup for incoming calls and sometimes the audio of the recording works great. At times though i can’t hear anything for 5-10 seconds and than it just jumps in the middle of the recording. The recording file used to be a WAV file but since my upload speed isn’t that great I converted it (sox) to a GSM file.
In the CLI I can see that the file is being played so it’s not a matter of ownership and the fact that sometimes audio is being played correctly indicates that the file should be OK.
What do you guys think? Help is very much appreciated!
What type of trunking are you using? Perhaps SIP?
Yes it is a SIP trunk