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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>6 seconds here. Though I could swear when I tried it earlier it was even longer than that.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>-- Nathan<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> VoiceOps [mailto:voiceops-bounces@voiceops.org] <b>On Behalf Of </b>Carlos Alvarez<br><b>Sent:</b> Wednesday, December 02, 2015 7:07 PM<br><b>To:</b> voiceops@voiceops.org<br><b>Subject:</b> Re: [VoiceOps] SIP provider that will route to free conference services<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>I made three calls with PDD maybe 2-3 seconds on Onvoy. <br><br>Sent from my iPhone<o:p></o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br>On Dec 2, 2015, at 7:21 PM, Nathan Anderson <<a href="mailto:nathana@fsr.com">nathana@fsr.com</a>> wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>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...</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>--</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Nathan Anderson</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>First Step Internet, LLC</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><a href="mailto:nathana@fsr.com">nathana@fsr.com</a></span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> VoiceOps [<a href="mailto:voiceops-bounces@voiceops.org">mailto:voiceops-bounces@voiceops.org</a>] <b>On Behalf Of </b>Evan P. Hall<br><b>Sent:</b> Wednesday, December 02, 2015 1:31 PM<br><b>To:</b> Carlos Alvarez<br><b>Cc:</b> <a href="mailto:voiceops@voiceops.org">voiceops@voiceops.org</a><br><b>Subject:</b> Re: [VoiceOps] SIP provider that will route to free conference services</span><o:p></o:p></p></div></div><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>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.</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Here is one of the phone numbers: 712-775-7031. 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.</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>-Evan</span><o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>From:</span></b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> VoiceOps [<a href="mailto:voiceops-bounces@voiceops.org">mailto:voiceops-bounces@voiceops.org</a>] <b>On Behalf Of </b>Carlos Alvarez<br><b>Sent:</b> Wednesday, December 2, 2015 12:56 PM<br><b>To:</b> <a href="mailto:voiceops@voiceops.org">voiceops@voiceops.org</a><br><b>Subject:</b> Re: [VoiceOps] SIP provider that will route to free conference services</span><o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><div><p class=MsoNormal>And then, please do record that phone call!<o:p></o:p></p><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal>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.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div></div><div><p class=MsoNormal> <o:p></o:p></p><div><p class=MsoNormal>On Wed, Dec 2, 2015 at 1:52 PM, Alex Balashov <<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.com</a>> wrote:<o:p></o:p></p><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'><p class=MsoNormal style='margin-bottom:12.0pt'>On 12/02/2015 03:49 PM, Evan P. Hall wrote:<o:p></o:p></p><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'><p class=MsoNormal>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.<o:p></o:p></p></blockquote><p class=MsoNormal><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" target="_blank">https://www.fcc.gov/document/fcc-releases-order-combating-rural-call-completion-problems</a><span style='color:#888888'><br><br><span class=hoenzb>-- Alex</span><br><br><span class=hoenzb>-- </span><br><span class=hoenzb>Alex Balashov | Principal | Evariste Systems LLC</span><br><span class=hoenzb>303 Perimeter Center North, Suite 300</span><br><span class=hoenzb>Atlanta, GA 30346</span><br><span class=hoenzb>United States</span><br><br><span class=hoenzb>Tel: <a href="tel:%2B1-800-250-5920" target="_blank">+1-800-250-5920</a> (toll-free) / <a href="tel:%2B1-678-954-0671" target="_blank">+1-678-954-0671</a> (direct)</span><br><span class=hoenzb>Web: <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a>, <a href="http://www.csrpswitch.com/" target="_blank">http://www.csrpswitch.com/</a></span></span><o:p></o:p></p><div><div><p class=MsoNormal><br>_______________________________________________<br>VoiceOps mailing list<br><a href="mailto:VoiceOps@voiceops.org" target="_blank">VoiceOps@voiceops.org</a><br><a href="https://puck.nether.net/mailman/listinfo/voiceops" target="_blank">https://puck.nether.net/mailman/listinfo/voiceops</a><o:p></o:p></p></div></div></blockquote></div><p class=MsoNormal> <o:p></o:p></p></div></div></blockquote></div></body></html>