VAD on FreePBX

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?

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?

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.

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.

I checked my older system and found the same issue! Searching turned up . 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.

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I will give that a try thanks.

it did work thanks for the help

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