IVR recordings not playing, but logs say they are playing

I’m working to setup a new Freepbx distro install, and am getting strange behavior when IRV recording is supposed to play. The log files says it’s playing, but I hear nothing when I call in. I’m running latest FreePBX distro ISO and I chose asterisk 11.

Here’s what the log says:

pbx.c: – Executing [[email protected]:9] Set(“SIP/Flowroute-00000003”, “IVR_MSG=custom/after_hours”) in new stack
pbx.c: – Executing [[email protected]:10] Set(“SIP/Flowroute-00000003”, “TIMEOUT(digit)=3”) in new stack
func_timeout.c: – Digit timeout set to 3.000
pbx.c: – Executing [[email protected]:11] ExecIf(“SIP/Flowroute-00000003”, “1?Background(custom/after_hours)”) in new stack
file.c: – <SIP/Flowroute-00000003> Playing ‘custom/after_hours.slin’ (language ‘en’)

It says it’s playing my “after_hours” message, but I’m hearing nothing when I call in. Moreover, the log is saying it is playing “custom/after_hours.slin” The file is actually after_hours.wav. Does freepbx do some kind of convert on the wave file?

I’m not sure why this isn’t working.

Thanks,

Greg

Greg: How did you record the wav files? If you didn’t just use system recording via an extension then it could be that the format of the recording is incorrect. The system recording screen states: “Note that if you’re using .wav, (eg, recorded with Microsoft Recorder) the file must be PCM Encoded, 16 Bits, at 8000Hz”

I think I might have a firewall issue. After further investigation, I found that I’m not getting any audio from the pbx. I’ve been testing this install with a low cost SIP trunk, which may not play well with my firewall. I’m going to switch to my primary SIP trunk, which I know how to configure with my firewall. I’ll report back if I’m still having a problem after getting the other trunk setup.

Thanks!

The problem did turn out to be the cheap SIP provider: Flowroute. I added my primary SIP trunk (VoicePulse) to FreePBX install and all works fine. The problem is that Flowroute doesn’t play well with my firewall/UTM, which is Sophos UTM 9.2. I tried a bunch of different port forwards, firewall rules, and even SIP helper in Sophos UTM, but I couldn’t get sound to be transmitted out of the firewall nor could I make any outgoing calls. Incoming calls would ring and extension, but once I picked up it was dead air.

Moral of the story is to be careful with the cheap SIP providers.

Greg