VAD on FreePBX

FreePBX
1

I have been having a problem with T-Mobile calls drop after 20 sec of voice-mail. My sip trunk provider COX talked with them about it. Net result T-Mobile want me to enable some type of VAD or comfort noise generation. From searching Google it looks like asterisk dosn’t support this, is that correct? Any suggestion?

2

If anything, it sounds like VAD could cause such a problem. But please be more specific:

  1. A call from your extension to a T-Mobile customer, leaving a voicemail, call drops after 20 seconds of recording?
  2. A call from your extension to retrieve your T-Mo voicemail, call drops after 20 seconds of playback?
  3. A call from a T-Mo customer into your PBX, goes to voicemail, call drops after 20 seconds of recording?
  4. A call from your T-Mo line to retrieve your FreePBX voicemail, call drops after 20 seconds of playback?

VAD at your end does not seem relevant for (1).
If (2) or (4), does speaking a word every few seconds during playback avoid the drop?
If (3), can you confirm e.g. with tcpdump that the PBX is sending valid RTP while the caller is leaving a message?

Does your PBX connect directly to Cox or do you have an SBC provided by them? What router and/or firewall do you have between the PBX and Cox?

3

Sorry it’s case 3. We get calls from T-Mobile, to our voice-mail and 20 sec into the recording M-Mobile drop the call.

I have been giving COX time, and phone number for when we get the cut off voice mails, for about 3 months now. The guy at COX got the T-Mobile people to look into it, and that is what they came up it. The calls don’t drop if you talk to people on T-Mobile only when they go to voice mail.

Cox has a box in our data center that the PBX connect to. The PBX is a virtual box running on VMware, and the switches are Cisco configured for QoS for a voice vlan.

It’s a older versions of FreePBX 11.

Not sure how to do a tcpdump, to do that check.

4

I have never employed it for this use case, but the RTP Keep Alive parameter in Settings, Asterisk SIP Settings, Chan_SIP Settings might keep the session alive for you. Set it to less than whatever T-Mobile’s disconnect limit is.

5

I checked my older system and found the same issue! Searching turned up https://thecomputerperson.wordpress.com/2017/05/10/asterisk-voicemail-hangs-up-on-callers-after-a-few-seconds/ . Looking at my asterisk.conf, I noticed
;transmit_silence_during_record = yes ; Transmit SLINEAR silence while a channel is being recorded
so uncommented that (and the [options] header), restarted Asterisk and problem is gone.

I didn’t try Lorne’s fix (which should use less resources), because I’d rather have ‘normal looking’ RTP for troubleshooting purposes and my lightly used system has enough horsepower.

1 Like
6

I will give that a try thanks.

edit
it did work thanks for the help

7

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