Summary:	ASTERISK-29889: asterisk.conf transmit_silence does not work in VoiceMail()
Reporter:	Luke Escude (lukeescude)	Labels:
Date Opened:	2022-01-31 12:08:12.000-0600	Date Closed:	2022-01-31 12:18:52.000-0600
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Applications/app_voicemail
Versions:	16.22.0	Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	Lately, AT&T has been dropping calls to our voicemail system because their SIP proxies aren't receiving audio from Asterisk when the inbound caller is leaving a voicemail. We need to simulate silence (real audio, not comfort noise) on the upstream channel while the inbound caller is leaving a message so AT&T doesn't cut off the call. I believe transmit_silence is supposed to enable this, but it does not seem to be working. Do we have to restart Asterisk in order for transmit_silence to take effect? Or is there a bug here? Steps to Recreate: 1. Route an inbound DID to Voicemail() 2. Call into that DID with an AT&T cell phone 3. Leave a longer message, like 30 seconds 4. AT&T caller's side will cut off the call about 15-20 seconds in, with RTP Timeout cited in the BYE. Steps to Diagnose: 1. When you're on the call, run pjsip show channelstats periodically and you will notice Asterisk is not sending proper RTP packets anymore.
Comments:	By: Asterisk Team (asteriskteam) 2022-01-31 12:08:12.788-0600 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Luke Escude (lukeescude) 2022-01-31 12:16:03.078-0600 "core show settings" shows the following: Transmit silence during rec: Disabled So I think Asterisk must be restarted for this to take effect. core reload may not be enough. By: Luke Escude (lukeescude) 2022-01-31 12:18:41.623-0600 Yep here is someone else who ran into this as well: https://community.asterisk.org/t/voicemail-rtp-issue/83745/7 Closing the ticket now, sorry.