[asterisk-users] asterisk-users Digest, Vol 151, Issue 23

Saint Michael venefax at gmail.com
Wed Feb 22 12:14:29 CST 2017


Theory: The carrier is not responding with 100 Trying in the expected time.
Hence, Asterisk is sending the INVITE again.

On Wed, Feb 22, 2017 at 1:00 PM, <asterisk-users-request at lists.digium.com>
wrote:

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> Today's Topics:
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>    1. Looking for Speech Recognition (ASR) suggestions (Dan Cropp)
>    2. multiple outbound invites (Jeff LaCoursiere)
>
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> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 22 Feb 2017 15:43:56 +0000
> From: Dan Cropp <dan at amtelco.com>
> To: "asterisk-users at lists.digium.com"
>         <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Looking for Speech Recognition (ASR)
>         suggestions
> Message-ID:
>         <41223E927281D842A48CC18032B36CC30118FAFEC8 at mail2010c.amtelco.com>
> Content-Type: text/plain; charset="us-ascii"
>
> 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
> correct?
>
> Have a great day!
>
> Dan
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> ------------------------------
>
> Message: 2
> Date: Wed, 22 Feb 2017 11:57:16 -0600
> From: Jeff LaCoursiere <jeff at jeff.net>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] multiple outbound invites
> Message-ID: <a38a9ce9-b53d-bbef-5897-422ef7cd4d38 at jeff.net>
> Content-Type: text/plain; charset=utf-8; format=flowed
>
>
> 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 few days ago ALL calls sent to the 'primary' were returned as
> busy, though the secondary terminated them fine.  We have a balance, and
> funny enough international calls are going through fine, just not US
> calls.  I opened a ticket.
>
> The response form the carrier is that our asterisk is sending four
> simultaneous invites within one second, and for that reason the call is
> rejected.
>
> I did a packet trace and was able to confirm this is true - only US
> calls sent to this carrier cause our end to send four identical
> simultaneous invites.  When it fails, a single invite for the same call
> is sent to the secondary, which is terminated without issue.
>
> Happy to send the SIP trace if any would care to see it, but is there a
> reason anyone can think of that our asterisk (11.11.0) would suddenly
> start doing this?  It may be that it has been doing it all along, and
> our carrier just started rejected calls that come in this way, I'm not
> sure.
>
> Cheers,
>
> j
>
>
>
>
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> End of asterisk-users Digest, Vol 151, Issue 23
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