Voicemail drops after 20 seconds (with some mobile phones)

[glasses685]

Hi all, 

Was just wondering if anyone had a suggestion for this issue. I run an older version of UniteMe, that sends calls over an ATT SIP Trunk (with a Sangoma SBC in the middle). 

The issue is when inbound callers on some mobile phones call to leave a voicemail, and IVR picks up, the call will drop after 20 seconds. This only happens with certain mobile phones and does not happen on active calls. It only started recently.

According to AT&T it's a known issue with VAD causing problems with some carriers and it needs to be disabled (not sure how to do that). After talking to Sangoma my understanding is that some providers (T-Mobile?) aren't accepting comfort noise packets and are just dropping the call after 20 seconds of (perceived) silence. We tried setting the suppress_cng variable to true but the issue persisted because of (according to them) lack of RTP from the UniteMe server after voicemail picks up, even after disabling comfort noise. 

I can attach a PCAP if it'd be helpful - just wanted to see if anyone had any general deas on how to fix this. I'm guessing some carrier must have changed a setting recently. Thanks! 

[Support]

Are you by any chance using a compressed codec? If so switching to PCMU may help.

--
You received this message because you are subscribed to the Google Groups "sipxcom-users" group.
To unsubscribe from this group and stop receiving emails from it, send an email to sipxcom-user...@googlegroups.com.
To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/d401c048-394c-4f18-93f3-09b0f939a227n%40googlegroups.com.

[glasses685]

Thanks for the input! In this case, we're not. I'm suspecting the issue is this - https://wiki.kolmisoft.com/index.php/Voicemail_call_is_cut_by_Originator_due_to_lack_of_RTP/audio 

However I'm not sure how or if I can adjust either of those on my system as it's the UniteMe version of Asterisk. 

[Support]

You may check whether your trunk side SDP has the a=rtp-keepalive setting enabled.

The built in sipX SBC has a media keep alive timer setting, but it doesn’t look like you are using it, would expect Sangoma’s SBC to have something similar.

There is also a Media Services RTP keep alive setting, but it is for the opposite scenario where Freeswitch disconnects after a silent period.

To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/3357889d-0fd4-432f-a3a2-3a4026802fb4n%40googlegroups.com.

[Todd Hodgen]

I've seen this with a cell carrier that was arriving with low hops.  We ended up having the SBC increas hops and fixed it.  I would look at what the call arives with, and compare to a call that doesn't drop.

Sent with my two left twiddling thumbs

---

From: sipxco...@googlegroups.com <sipxco...@googlegroups.com> on behalf of Support <sup...@onrelay.net>
Sent: Tuesday, November 18, 2025 3:18:17 PM
To: glasses685 <tec...@gmail.com>
Cc: sipxcom-users <sipxco...@googlegroups.com>
Subject: Re: [sipxcom-users] Voicemail drops after 20 seconds (with some mobile phones)

 

To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/68CA34CF-E802-4435-9926-3A2E15218F00%40onrelay.net.

[glasses685]

Thanks for the suggestions, I really appreciate it - I'll see if there's some way ATT can add this keep alive setting or if increasing the hop count would help.  Anyone know if either of these settings (rtp keepalive or transmit_silence) are possible to add in UniteMe? I don't see an option in the UI but I could certainly edit a conf file if it would work. 

[glasses685]

Wondering if this might be a viable fix - https://docs.coredial.com/en/articles/1104-custom-freeswitch-allowing-voicemails-longer-than-60-seconds-to-be-left-from-external-callers 

Although I'll need to set it lower than the default I think as I'm having calls drop after 20 seconds.

[glasses685]

Hey all, what seems to have resolved it (for me) as enabling this setting - 
https://developer.signalwire.com/freeswitch/Channel-Variables-Catalog/bridge_generate_comfort_noise_16353601/

Basically I just edited the inbound call routing rule on the SBC to set that to true - might be that I needed that one since we use the "Transfer by Bridging the Call" setting. Apparently some carriers will drop the call if they go more than 30 seconds without seeing any RTP from sipxcom, this seems to prevent that. I appreciate all the input. Might be something to look out for as it only became an issue recently, I can only guess that some carrier changed their setting.

[Support]

Thank you for bottoming that out!

sipX already has the option in the built in SBC to manage this (see below), both method and interval.

However, as it is a straightforward admin setting fix for those that use other SBCs we have also added an issue to add the below freeswitch setting as a media services option. See https://github.com/onrelay/sipxecs/issues/51.

To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/2f9fc93f-c5a2-4030-89bd-953ec894bec7n%40googlegroups.com.