search for: strommen

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2018 Jan 20
2
Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
On 20 January 2018 at 23:30, Tim S <tim.strommen at gmail.com> wrote: > I have seen this take over 2 seconds before on a sluggish machine. Thanks - my host uses SSD and everything seems pretty quick, but I'll give it a 1 second pause. > you'd need to pipe that to a Google Speech API tunnel. > That's probably not somethi...
2018 Jan 20
2
Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
On Saturday 20 January 2018 at 18:45:49, Jonathan H wrote: > Oh, what a good idea! That's exactly the kind of lateral thinking I > was hoping someone would come up with. > > I thought it was called MixMonitor, and tried to wrap my head around > it but couldn't. MixMonitor is related, but different (and as the name suggests, automatically mixes the two channels, so I
2018 Jan 20
2
Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
Just a quick and dirty thought, try the MONITOR application. Pseudo-code: Anchor-point PLAYBACK ("press or say") MONITOR (use the split audio files mode, not the mixed - this way you can roughly separate which side did the "talking") READ (audio file "1 to 5", try to grab one digit) STOPMONITOR IF (READ variable timed-out, send the incoming half of the monitor file
2017 Jun 15
3
OT: Explain where mailing list bouncing comes from ?
Another "me too" (also Gmail). I just received my 4th "account suspended, too many bounces" email, after having several days of lost mailing list content over a short vacation break the last time. When I notified the admin email account of the failure, it seemed the responder missed the point about the emails, saying the link had expired (it had been more than three days
2017 May 08
2
Call does not go voicemail
The "error" I was talking about was in your log: "...== Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-6364'..." The call terminated here in a error which prevented the dialplan from continuing. Something there is broken, my recommendation is to check you registrations first inside asterisk: > sip show peers Something wasn't