<div dir="ltr"><div class="gmail_default" style="font-size:small">Theory: The carrier is not responding with 100 Trying in the expected time. Hence, Asterisk is sending the INVITE again.<br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Feb 22, 2017 at 1:00 PM, <span dir="ltr"><<a href="mailto:asterisk-users-request@lists.digium.com" target="_blank">asterisk-users-request@lists.digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Send asterisk-users mailing list submissions to<br>
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Today's Topics:<br>
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1. Looking for Speech Recognition (ASR) suggestions (Dan Cropp)<br>
2. multiple outbound invites (Jeff LaCoursiere)<br>
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Message: 1<br>
Date: Wed, 22 Feb 2017 15:43:56 +0000<br>
From: Dan Cropp <<a href="mailto:dan@amtelco.com">dan@amtelco.com</a>><br>
To: "<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.<wbr>com</a>"<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.<wbr>com</a>><br>
Subject: [asterisk-users] Looking for Speech Recognition (ASR)<br>
suggestions<br>
Message-ID:<br>
<<a href="mailto:41223E927281D842A48CC18032B36CC30118FAFEC8@mail2010c.amtelco.com">41223E927281D842A48CC18032B36<wbr>CC30118FAFEC8@mail2010c.<wbr>amtelco.com</a>><br>
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<br>
Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS?<br>
<br>
Could anyone provide pros/cons for the various ASR options for Asterisk?<br>
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?<br>
<br>
Have a great day!<br>
<br>
Dan<br>
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Message: 2<br>
Date: Wed, 22 Feb 2017 11:57:16 -0600<br>
From: Jeff LaCoursiere <<a href="mailto:jeff@jeff.net">jeff@jeff.net</a>><br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.<wbr>com</a>><br>
Subject: [asterisk-users] multiple outbound invites<br>
Message-ID: <<a href="mailto:a38a9ce9-b53d-bbef-5897-422ef7cd4d38@jeff.net">a38a9ce9-b53d-bbef-5897-<wbr>422ef7cd4d38@jeff.net</a>><br>
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Hello,<br>
<br>
I have two upstream providers we use for US termination. The dialplan<br>
sends calls out the "primary" and if that fails for specific reasons, it<br>
sends the same call out the "secondary". This has worked well for us<br>
when we are lazy about keeping balances up, for example.<br>
<br>
Starting a few days ago ALL calls sent to the 'primary' were returned as<br>
busy, though the secondary terminated them fine. We have a balance, and<br>
funny enough international calls are going through fine, just not US<br>
calls. I opened a ticket.<br>
<br>
The response form the carrier is that our asterisk is sending four<br>
simultaneous invites within one second, and for that reason the call is<br>
rejected.<br>
<br>
I did a packet trace and was able to confirm this is true - only US<br>
calls sent to this carrier cause our end to send four identical<br>
simultaneous invites. When it fails, a single invite for the same call<br>
is sent to the secondary, which is terminated without issue.<br>
<br>
Happy to send the SIP trace if any would care to see it, but is there a<br>
reason anyone can think of that our asterisk (11.11.0) would suddenly<br>
start doing this? It may be that it has been doing it all along, and<br>
our carrier just started rejected calls that come in this way, I'm not sure.<br>
<br>
Cheers,<br>
<br>
j<br>
<br>
<br>
<br>
<br>
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</blockquote></div><br></div>