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

[Luke Escude (JIRA)]

Luke Escude created ASTERISK-29889:
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             Summary: 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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