I recently began administering FreePBX at my workplace, and we are experiencing intermittent issues with inbound calls selecting extensions from the menu.
When someone calls in and tries to select an option from the menu, one of three things usually happens:
When selecting the desired extension from the menu, nothing happens. For example: If someone selects ‘1’, the menu keeps going on as if nothing was pressed. If they keep pressing it, it will eventually bring them to the extension.
When selecting the extension, a recording states “I’m sorry, that is not a valid extension. Please try again.” Upon trying again, it will either repeat the message, or go through to the extension.
It works. When someone selects an option, it goes right to that extension.
Any ideas on what I should look at? What information could I share to help get to the bottom of this?
Here’s the IVR info:
IVR DTMF Options:
Enable direct dial: Enabled
Alert Info: None
Ringer Volume Override: None
Invalid Retries: 3
Invalid Retry Recording: Default
Append Announcement to Invalid: No
Return on Invalid: Yes
Invalid Recording: Default
Invalid Destination: IVR (wdrt-main)
Timeout Retries: 3
Timeout Retry Recording: Default
Append Announcement on Timeout: No
Return on Timeout: Yes
Timeout Destination: IVR (wdrt-main)
Return to IVR after VM: No
1 - Voicemail Blasting (Community Comment Line <1>)
2 - Ring Groups (200 Studios)
3 - Extensions (120)
4 - Extensions (140)
5 - Extensions (130)
6 - Voicemail Blasting (Staff VM Blast <105>)
9 - IVR (wdrt-main)
I suspect your system is not reliably detecting DTMF. You can confirm this by browsing to Settings, Asterisk Logfiles and enabling DTMF logging for both console and full log, and then grepping for a call trace after the fact:
I did a test call and followed up w/the grep command. Here’s what I got:
:[2020-03-30 10:16:45] VERBOSE[C-000009bb] pbx.c: Executing [[email protected]:1] Set(“DAHDI/i1/6086387064-94b”, “TOUCH_MONITOR=1585581379.7860”) in new stack
I then did ‘grep C-000009bb /var/log/asterisk/full-20200330’ and got nothing back.
OK, I tried the ‘grep C-’ command on another call, and got a lot of info. pastebin.freepbx.org/view/145a55cf#L5
The whole point of sharing call traces via pastebin is to prevent flooding the forum with unreadable lines. You should be able to post links now.
You haven’t enabled DTMF logging.
If the forum does not permit posting links, please post something that is easy to select, requires no editing by the reader, and can be opened with a right-click option on most browsers. For example:
Your trust level has been upgraded @WDRT, you should be able to link now.
Thanks! Here is the link: pastebin.freepbx.org/view/145a55cf
I re-checked, and DTMF logging is ‘On’ for both console and full.
Also, not sure if it’s relevant, but we are using Asterisk version 14.7.5.
I also have been testing calling extensions internally. When I dial our number from an internal phone, I can select any extension at any time during the IVR menu, and it works consistently. The issue is isolated to incoming calls only.
I placed another inbound test call, and here is what I found under the ‘Asterisk Logfiles’ tab:
[2020-03-31 12:05:56] DTMF[C-000009db] channel.c: DTMF begin ‘1’ received on DAHDI/i1/6086387064-96a
[2020-03-31 12:05:56] DTMF[C-000009db] channel.c: DTMF begin ignored ‘1’ on DAHDI/i1/6086387064-96a
[2020-03-31 12:05:56] WARNING res_musiconhold.c: poll() failed: Interrupted system call
[2020-03-31 12:05:56] DTMF[C-000009db] channel.c: DTMF end ‘1’ received on DAHDI/i1/6086387064-96a, duration 25 ms
[2020-03-31 12:05:56] DTMF[C-000009db] channel.c: DTMF end passthrough ‘1’ on DAHDI/i1/6086387064-96a
I have also isolated the problem to selecting options 1, 2 or 3 in the IVR. 4, 5, 6 and 9 all seem to work consistently, and can be selected at any time successfully. 1, 2, and 3 consistently do not work when selected, and after selecting the number multiple times, either begin to work or the “I’m sorry, that is not a valid extension” message plays.
Update your PBX and set the IVR Force Strict Dial Timeout to No - Legacy.
Also, I would move Asterisk to a supported version
Thanks! I will work on getting this done and check back in.
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