Hello,
I have been scouring the web for a solution to an issue with my PBX, there is no audio from my voice files on my already setup IVR,

Solutions tried:
Changed ownership of sounds folder, used a sample audio file from the system (gsm format), dabbled into echo cancellation in the zap* file, tested IVR via an extension (7777) and it all works - Audio plays with DTMF selection, but once i dial in from my sip trunk i get dead air. Asterisk verbose and debug show file playing as should (so no errors would be picked within the log or onscreen dump). I am currently using Elastix 2.5 (updated all repos via yum), Asterisk 11.17 on a Centos 6 server. I truly am stumped. Codecs are set to allow=all by the way. All i ask is to be educated Please

Hello all.. Thanks for having me...
Marudi, based in Abuja, Nigeria. Quick learner (eager too, soooo much to know and learn :D), loads of ideas and experiences and i can only hope to be able to help someone out when the time comes. humble

@scottalanmiller Yes... Everything works fine when I dialed in and pointed inbound route to an extension, rings and audio is perfect. Web interface, logs, Asterisk functions, modules loaded everything works fine. This is what confuses me the most. Unfortunately I cant just erase the entire server and start from scratch.

Hello all.. Thanks for having me...
Marudi, based in Abuja, Nigeria. Quick learner (eager too, soooo much to know and learn :D), loads of ideas and experiences and i can only hope to be able to help someone out when the time comes. humble

@scottalanmiller Yes... Everything works fine when I dialed in and pointed inbound route to an extension, rings and audio is perfect. Web interface, logs, Asterisk functions, modules loaded everything works fine. This is what confuses me the most. Unfortunately I cant just erase the entire server and start from scratch.

Hello,
I have been scouring the web for a solution to an issue with my PBX, there is no audio from my voice files on my already setup IVR,

Solutions tried:
Changed ownership of sounds folder, used a sample audio file from the system (gsm format), dabbled into echo cancellation in the zap* file, tested IVR via an extension (7777) and it all works - Audio plays with DTMF selection, but once i dial in from my sip trunk i get dead air. Asterisk verbose and debug show file playing as should (so no errors would be picked within the log or onscreen dump). I am currently using Elastix 2.5 (updated all repos via yum), Asterisk 11.17 on a Centos 6 server. I truly am stumped. Codecs are set to allow=all by the way. All i ask is to be educated Please