Hi,
that could be caused when your upstream offers "100rel" and your Asterisk does
not get a response fast enough from your upstream.
Is your outbound peer monitored by the qualify feature (qualify=yes)?
Then asterisk should calculate the round-trip-time until a response arrives and
should not
Maybe your firewall is blocking receiving packets from that provider or some
sip helper is messing the returning packets so asterisk is not recieving a
response and resending the invite
Original Message
From: j...@jeff.net
Sent: February 22, 2017 7:57 PM
To: asterisk-users@lists.digium.com
se of this, my initial thought is Nuance or Lumenvox. Does this sound
> correct?
>
> Have a great day!
>
> Dan
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Hello,
I have two upstream providers we use for US termination. The dialplan
sends calls out the "primary" and if that fails for specific reasons, it
sends the same call out the "secondary". This has worked well for us
when we are lazy about keeping balances up, for example.
Starting a
Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS?
Could anyone provide pros/cons for the various ASR options for Asterisk?
We need the ability for very large grammars (over 100,000 options). Because of
this, my initial thought is Nuance or Lumenvox. Does this sound
