<div dir="ltr">that's because <span style="color:rgb(31,73,125);font-family:Calibri,sans-serif;font-size:14.6667px"> </span><a href="tel:712-775-7031" value="+17127757031" target="_blank" style="font-family:Calibri,sans-serif;font-size:14.6667px">712-775</a> is an expensive rate.</div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Dec 2, 2015 at 10:14 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I would guess the PDD is upstream LCR hunting through a large amount of vendors who fail to take the call before finding one that will take it.<br>
<br>
<span class="">--<br>
Alex Balashov | Principal | Evariste Systems LLC<br>
303 Perimeter Center North, Suite 300<br>
Atlanta, GA 30346<br>
United States<br>
<br>
Tel: <a href="tel:%2B1-800-250-5920" value="+18002505920">+1-800-250-5920</a> (toll-free) / <a href="tel:%2B1-678-954-0671" value="+16789540671">+1-678-954-0671</a> (direct)<br>
Web: <a href="http://www.evaristesys.com/" rel="noreferrer" target="_blank">http://www.evaristesys.com/</a>, <a href="http://www.csrpswitch.com/" rel="noreferrer" target="_blank">http://www.csrpswitch.com/</a><br>
<br>
</span>Sent from my BlackBerry.<br>
Original Message <br>
From: Nathan Anderson<br>
Sent: Wednesday, December 2, 2015 22:12<br>
To: 'Carlos Alvarez'; <a href="mailto:voiceops@voiceops.org">voiceops@voiceops.org</a><br>
<div class="HOEnZb"><div class="h5">Subject: Re: [VoiceOps] SIP provider that will route to free conference services<br>
<br>
6 seconds here. Though I could swear when I tried it earlier it was even longer than that.<br>
<br>
-- Nathan<br>
<br>
From: VoiceOps [mailto:<a href="mailto:voiceops-bounces@voiceops.org">voiceops-bounces@voiceops.org</a>] On Behalf Of Carlos Alvarez<br>
Sent: Wednesday, December 02, 2015 7:07 PM<br>
To: <a href="mailto:voiceops@voiceops.org">voiceops@voiceops.org</a><br>
Subject: Re: [VoiceOps] SIP provider that will route to free conference services<br>
<br>
I made three calls with PDD maybe 2-3 seconds on Onvoy.<br>
<br>
Sent from my iPhone<br>
<br>
On Dec 2, 2015, at 7:21 PM, Nathan Anderson <<a href="mailto:nathana@fsr.com">nathana@fsr.com</a>> wrote:<br>
<br>
That number seems to work for me from Flowroute. But it *does* take an awful long time to connect, like they are trying to go down through a list of carriers in their LCR table and keep getting rejected, until it finally manages to go through on a carrier that is a fair ways down the list...<br>
<br>
--<br>
Nathan Anderson<br>
First Step Internet, LLC<br>
<a href="mailto:nathana@fsr.com">nathana@fsr.com</a><br>
<br>
From: VoiceOps [mailto:<a href="mailto:voiceops-bounces@voiceops.org">voiceops-bounces@voiceops.org</a>] On Behalf Of Evan P. Hall<br>
Sent: Wednesday, December 02, 2015 1:31 PM<br>
To: Carlos Alvarez<br>
Cc: <a href="mailto:voiceops@voiceops.org">voiceops@voiceops.org</a><br>
Subject: Re: [VoiceOps] SIP provider that will route to free conference services<br>
<br>
We do offer conferencing, but these customers are needing to join large conferences arranged by business partners or vendors and they are not able to influence the service used.<br>
<br>
Here is one of the phone numbers: <a href="tel:712-775-7031" value="+17127757031">712-775-7031</a>. I have a carrier that rejects it outright and 2 that it just rings and rings and never answers. If I call from my cell phone, it immediately answers and prompts for conference information.<br>
<br>
-Evan<br>
<br>
From: VoiceOps [mailto:<a href="mailto:voiceops-bounces@voiceops.org">voiceops-bounces@voiceops.org</a>] On Behalf Of Carlos Alvarez<br>
Sent: Wednesday, December 2, 2015 12:56 PM<br>
To: <a href="mailto:voiceops@voiceops.org">voiceops@voiceops.org</a><br>
Subject: Re: [VoiceOps] SIP provider that will route to free conference services<br>
<br>
And then, please do record that phone call!<br>
<br>
If you give me a sample number to test, I'll try it on our carriers. But on the other hand, why not just provide the conferencing service on your own service? That's what we do; we've simply told our customers not to use them and give them conferencing ability as part of our services.<br>
<br>
<br>
On Wed, Dec 2, 2015 at 1:52 PM, Alex Balashov <<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>> wrote:<br>
On 12/02/2015 03:49 PM, Evan P. Hall wrote:<br>
<br>
We have a few customers who need to reach free conference call<br>
services and none of our current carriers will route the calls. Are<br>
there any providers out there that will take these calls via SIP? We<br>
are happy to pay a fair price per minute if somebody will take it.<br>
<br>
You could take, or threaten to take, the route of reporting your upstream termination carriers for noncompliance with the FCC's order on the subject. :-)<br>
<br>
<a href="https://www.fcc.gov/document/fcc-releases-order-combating-rural-call-completion-problems" rel="noreferrer" target="_blank">https://www.fcc.gov/document/fcc-releases-order-combating-rural-call-completion-problems</a><br>
<br>
-- Alex<br>
<br>
--<br>
Alex Balashov | Principal | Evariste Systems LLC<br>
303 Perimeter Center North, Suite 300<br>
Atlanta, GA 30346<br>
United States<br>
<br>
Tel: <a href="tel:%2B1-800-250-5920" value="+18002505920">+1-800-250-5920</a> (toll-free) / <a href="tel:%2B1-678-954-0671" value="+16789540671">+1-678-954-0671</a> (direct)<br>
Web: <a href="http://www.evaristesys.com/" rel="noreferrer" target="_blank">http://www.evaristesys.com/</a>, <a href="http://www.csrpswitch.com/" rel="noreferrer" target="_blank">http://www.csrpswitch.com/</a><br>
<br>
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</div></div></blockquote></div><br></div>