Audio Issues Troubleshooting Guide

Date: October 14, 2025 04:20 AM
Issues Reported:

*43 (Echo test) - No audio, just silence

*44 (Time) - Starts then ends with no audio

Voicemail - Incomplete prompts, numbers missing, call ends abruptly

Extension dialing during calls not working

45/46 queue prompts need verification

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🔍 Current Status

###enabled verbose logging (level 5)
✅ Enabled debug logging (level 3)
✅ Verified codecs: ulaw, alaw, gsm are enabled
✅ Verified sound files exist (738 files installed)
✅ Verified file permissions (asterisk:asterisk, 644)
✅ Verified dialplan loaded correctly
✅ Verified VoicemailMain application available

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🧪 Diagnostic Commands Run

Sound File Verification

# Checked directories - PASS
/usr/share/asterisk/sounds/digits/  - 90+ files
/usr/share/asterisk/sounds/vm-*     - 100+ voicemail prompts
/usr/share/asterisk/sounds/         - 738 total files
Permissions fixed

chmod 644 on all sound files
chmod 755 on directories
owner: asterisk:asterisk

Module Verification

# Echo application - LOADED
asterisk -rx "core show application Echo"
Result: Application available
Say applications - LOADED

asterisk -rx "core show applications" | grep Say
Result: SayNumber, SayDigits, SayUnixTime all available
Voicemail - LOADED

asterisk -rx "module show like voicemail"
Result: appvoicemailimap.so running

Endpoint/Codec Verification

# Extension 2001 codecs
Allowed: ulaw, alaw, gsm
RTP symmetric: Yes
Force rport: Yes
Direct media: No

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🎯 Likely Causes

Issue 1 & 2: 43/44 No Audio

Most likely cause: One-way RTP audio issue

Symptoms match:

• Call connects (so SIP signaling works)


• No audio heard (RTP not reaching client)


• Call works but silent

Possible reasons:


• Firewall blocking RTP - UDP ports 10000-20000 not open


• NAT/routing issue - RTP packets not reaching your device


• Codec negotiation - Client using codec Asterisk can't send


• STUN not working - External IP not properly discovered

Next steps to diagnose:


# While on a call to *43, check active channels
asterisk -rx "core show channels verbose"
Check RTP statistics

asterisk -rx "rtp show stats"
Check if RTP packets are being sent

(need to run during active call)

Issue 3: Voicemail Incomplete Prompts

Possible causes:


• Timeout too short - Call ending before prompts finish


• Missing sound files - Some vm-* files not found


• Codec issue - Same as above, can't hear prompts


• Voicemail config - Wrong paths or settings

Files verified present:


• vm-login.ulaw


• vm-password.ulaw


• vm-instructions.ulaw


• digits/0-9.ulaw


• Plus 100+ other vm-* files

Issue 4: Extension Dialing Not Enabled

Feature needed: Call transfers

Current status: Transfer features may not be enabled in features.conf

What's needed:


• Blind transfer: 2 or *


• Attended transfer: *3


• Or dial by extension feature

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🔧 Fixes Applied

1. Enabled Verbose Logging




asterisk -rx "core set verbose 5"
asterisk -rx "core set debug 3"

Now all calls will show detailed logs in /var/log/asterisk/messages

2. Fixed File Permissions




chmod 644 /usr/share/asterisk/sounds/*/
chmod 755 /usr/share/asterisk/sounds directories
chown asterisk:asterisk all sound files

3. Verified Dialplan


All feature codes properly loaded:

• *43 - Echo Test ✓


• *44 - Time Announcement ✓


• *45 - Queue Login ✓


• *46 - Queue Logout ✓


• *48 - Queue Status ✓


• *77 - MOH + Stats ✓


• *78 - MOH Preview ✓


• *97 - Voicemail ✓

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📋 Test Procedure

Test 1: Echo Test (*43)

What to do:


• Dial *43 from extension 2001


• Say something


• Listen for echo

Expected:


• Hear yourself with 0.5 second delay


• Clear audio quality


• Can talk for as long as you want

If still silent:


# Check logs during call
tail -f /var/log/asterisk/messages | grep -i "echo\|rtp\|audio"
Check if firewall allows RTP

netstat -tulpn | grep asterisk | grep -E "100[0-9]{2}"
Verify RTP ports open

sudo firewall-cmd --list-all | grep 10000-20000

Test 2: Time Announcement (*44)

What to do:


• Dial *44 from extension 2001


• Listen for time announcement

Expected:


• Hear: "Today is Sunday, October 14th, 2025. The time is 4:20 AM"


• Then call ends

If still silent:


# Check if SayUnixTime works
tail -f /var/log/asterisk/messages | grep -i "sayunixtime\|playback"

Test 3: Voicemail (*97)

What to do:


• Dial *97


• Follow prompts

Expected sequence:


• "Comedian mail" (intro)


• "Mailbox?" or auto-detect your mailbox


• "Password?" (if configured)


• Main menu: "You have X new messages..."


• Options menu with all choices

If still incomplete:


# Check voicemail context
asterisk -rx "voicemail show users"
Check sound file playback

tail -f /var/log/asterisk/messages | grep -i "vm-\|voicemail"

Test 4: Queue Prompts (45, 46)

Test *45 (Login):


• First dial - Should hear: "You are now logged into the support queue"


• Dial again - Should hear: "You are already logged into the queue"

Test *46 (Logout):


• First dial - Should hear: "You are now logged out of the support queue"


• Dial again - Should hear: "You are not currently logged into the queue"

Verify:


# Check which prompts are configured
grep -A 3 "exten => \*45" /etc/asterisk/extensions.conf
grep -A 3 "exten => \*46" /etc/asterisk/extensions.conf

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🚨 Real-Time Monitoring

During Your Next Test Call

Open SSH terminal and run:


# Terminal 1 - Watch messages log
tail -f /var/log/asterisk/messages
Terminal 2 - Watch channels

watch -n 1 'asterisk -rx "core show channels"'
Terminal 3 - Watch RTP

watch -n 1 'asterisk -rx "rtp show stats"'

Then dial *43 and note:


• Does channel show up?


• Does RTP show packets?


• Are there any WARNING or ERROR messages?


• Does codec show correctly?

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🔍 What I Need From You

Please Test and Report:

For *43 (Echo):


• [ ] Does call connect? (hear ringing/answered)


• [ ] Do you hear anything at all?


• [ ] Can you hear yourself echo back?


• [ ] How long before call ends?

For *44 (Time):


• [ ] Does call connect?


• [ ] Do you hear any audio?


• [ ] Does it say the time?


• [ ] Does call end immediately or after pause?

For *97 (Voicemail):


• [ ] What exactly do you hear?


• [ ] Where does it stop/end abruptly?


• [ ] Which menu options do you hear?


• [ ] What happens when you press digits?

For 45/46 (Queue):


• [ ] Which audio file actually plays?


• [ ] Is it the right one for the situation?


• [ ] Does it play completely?


• [ ] Any silence or cutoff?

Also Check:

On your phone/softphone:


• What codec is being used? (Check phone settings)


• Is STUN enabled?


• What's the audio codec priority?


• Any firewall on your computer?

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🔧 Additional Fixes to Try

If RTP/Audio Issues Persist

Option 1: Disable strictrtp temporarily


# Edit /etc/asterisk/rtp.conf
strictrtp=no
Reload

asterisk -rx "module reload resrtpasterisk.so"

Option 2: Check external IP


# Verify external IP is correct
asterisk -rx "pjsip show transports"
Should show:

transport-udp: 64.20.46.178:5060

transport-tailscale: 100.64.0.2:5060

Option 3: Test from different network


• Try calling from different WiFi


• Try calling from cellular data


• Try different SIP client

Option 4: Simplify codec list


# Test with only ulaw
Edit endpoint in pjsip.conf:

disallow=all
allow=ulaw
Reload

asterisk -rx "module reload res_pjsip.so"

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📊 Expected Log Output

Successful *43 Echo Test

[timestamp] VERBOSE[xxxxx] pbx.c: Executing [*43@flexpbx-internal:1] NoOp("PJSIP/2001-xxxxx", "Echo Test")
[timestamp] VERBOSE[xxxxx] pbx.c: Executing [*43@flexpbx-internal:2] Answer("PJSIP/2001-xxxxx", "")
[timestamp] VERBOSE[xxxxx] pbx.c: Executing [*43@flexpbx-internal:3] Wait("PJSIP/2001-xxxxx", "0.5")
[timestamp] VERBOSE[xxxxx] pbx.c: Executing [*43@flexpbx-internal:4] Echo("PJSIP/2001-xxxxx", "")
[timestamp] VERBOSE[xxxxx] app_echo.c: Echo media

If you see warnings/errors here, that's the problem!

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📞 Queue Prompts Configuration

Current Mapping (Verified Correct)

\*45 - Queue Login:


NOT in queue → Plays: callcueue-login.wav (5.8 sec)
ALREADY in queue → Plays: callcueue-loged-in-to-out-prompt.wav (8.3 sec)

\*46 - Queue Logout:


IN queue → Plays: callcueue-logout.wav (7.5 sec)
NOT in queue → Plays: callcueue-loged-out-to-in-prompt.wav (11.0 sec)

Files exist in:


• /usr/share/asterisk/sounds/callcueue-*.wav


• /usr/share/asterisk/sounds/callcueue-*.ulaw


• /usr/share/asterisk/sounds/callcueue-*.gsm

All 3 formats available for compatibility.

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✅ Action Items



• Test again with logging enabled


• Report back exactly what you hear (or don't hear)


• Check phone settings - codec, STUN, NAT settings


• Try from different device if possible


• Provide log snippets from /var/log/asterisk/messages during test call

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Status: Diagnostics enabled, awaiting test results
Next: Based on test results, we'll know if it's:


• RTP/firewall issue


• Codec issue


• Configuration issue


• Client-side issue

Logging: All enabled - will show detailed info on next call