[asterisk-bugs] [JIRA] (ASTERISK-29889) asterisk.conf transmit_silence does not work in VoiceMail()

[Luke Escude (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29889?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=257916#comment-257916 ] 

Luke Escude commented on ASTERISK-29889:
----------------------------------------

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.

> asterisk.conf transmit_silence does not work in VoiceMail()
> -----------------------------------------------------------
>
>                 Key: ASTERISK-29889
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29889
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_voicemail
>    Affects Versions: 16.22.0
>            Reporter: Luke Escude
>
> 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.



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