IVR not listening to options

Hello guys!

I am trying to see if anyone has experienced an issue with the IVR as I am experiencing, which is that when the announcement is playing on the IVR, the caller’s key presses are not taken until the end of the message.
I have enabled direct dial to see if that may have something to do with the issue, but that was a no go.

Thank you!

Direct dial allows the caller to directly dial the extension of the person they want, during the playing of the IVR.

Hi Red,

I know that it is used for dialing an extension during an announcement. It was something that I tried just in case. With that on, I also tried dialing the extension, to no avail.

Thank you!

You might try setting “signal ringing” in your inbound route to that IVR

Thank you Dicko!

The Inbound route sends the call to the Call Flow, which is then sent to a time condition in normal mode. It is the time condition that sends the call to the IVR. Does your suggestion apply even in this type of set up?

Additionally, the issue exists even in the sub-IVRs.

I have read somewhere about a delay in the IVR, where the phone system does not quickly acknowledge the options selected by the caller. Based on that, I figured that this delay could have been caused by, maybe, high CPU or high memory usage, so I started the process monitor and called in. As the IVR announcement started playing, the CPU spiked to 100% and the process that caused that was “Java” executed by the user “asterisk”. With the CPU, the memory spiked as well. As I was pressing the options, nothing happened, but when it dropped, I pressed the option again and it took it.

Could this be an issue, could it be a bug?

The hardware specs on the box are: CPU - Intel atom dual core @ 1.6GHz, with 2GB RAM, and 120GB SSD.

Thank you!

In effect, the call needs to be answered before any sdp audio will flow. “signal ringing” is in effect a sort of answer. The hardware is irrelevant,but sngrep will help you diagnose the sip and sdp traffic in any call

I will set the signal ringing and test it in the morning when the client opens their office.

Thank you!

In addition to signal ringing, you can also try turning on FAX detection on the line. This also turns on early audio processing, since the system has to listen to the call to decide where to send it.

I know it has been two weeks since I responded so here is an update.
Nothing worked yet, so I have purchased another computer (Intel i3 7500 CPU, ASUS motherboard, 120GB SSD, 8GB DDR4 RAM) and intalled Freepbx 14 with Asterisk 15. The only thing from the old setup is the Digium 8 port FXO card.
After setting it all up, recreating the extensions, the IVRs, announcements, Inbound, Outbound, etc., the issue still exists.
I enabled dahdi_monitor and watched for the audio levels as I was pressing the buttons…it showed that audio was present. To me that means that the tones are coming into the phone system, is that correct? I do have the audio files rx.raw and tx.raw.
The call flow is set up like this:
Call comes in > Call Flow > if Normal mode goes to Time Conditions > if during business hours > IVR with 3 options (0, 1, and 2)

if 0 goes to a Ring Group
if 1 goes to another IVR
if 2 goes to a Ring Group
I noticed that If I point the Time Condition to got to that “another IVR” none of my test calls failed when I was pressing the button making my selections, but then switching back to the original set up, it was failing again.
I turned off echo training, turned on fax detection, did not help.

Anything else I can try?

Thank you!

My suspicion is an echo canceler or doubletalk issue interfering with DTMF detection.

Set up a Misc Application that points to the problematic IVR. Dial the code from an extension on the system and see whether the IVR works properly.

Or, temporarily replace the IVR announcement with one containing some pauses. For example “For sales, press 0. [3 second pause] For technical support, press 1 [3 second pause] For customer service, press 2.” Then, see whether a key press during a pause is correctly heard.

Does your new system still max out the CPU during the IVR announcement?

How was the announcement created? Possibly, the system is having to transcode it or do other unwanted processing. If that’s happening, try converting it offline, e.g. with Audacity to a mono, linear PCM, 8 kHz sampling rate .wav file and uploading that. Also, don’t make it too loud (average level should be 10 to 13 dB below full scale) and don’t include any music or other unnecessary content.

Do you have any VoIP trunks on your system? If so, try routing a VoIP DID to the IVR and see whether that works any better.

I will try your suggestions and post back the findings.

The CPU no longer maxes out…at most it hits 7%.
As for the announcement, it was created through the voice memo on the iPhone and then I used Audacity to convert it to wav, but I don’t remember what sampling rate I chose, so I will convert the audio files again per your specs.

The POTS are analog from Comcast, no VoIP trunks, however, all 8 ports are taken so I could mess around with one of them to see what outcome I get.

Thank you!

In the system recording options, there are several pieces that might help you. The most obvious is using an extension to record the files, the other is to use the “transcoding” options at the bottom of the page to make sure the format of the file is correct.

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