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<p class="MsoNormal">14400 works nicely.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Until it doesn’t.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">And then you use 9600.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><b>From:</b> cisco-voip <cisco-voip-bounces@puck.nether.net>
<b>On Behalf Of </b>Brian Meade<br>
<b>Sent:</b> Tuesday, May 8, 2018 2:24 PM<br>
<b>To:</b> Anthony Holloway <avholloway+cisco-voip@gmail.com><br>
<b>Cc:</b> Adrian Arevalo-Orozco <adrian.arevalo.orozco@chemeketa.edu>; cisco-voip@puck.nether.net; Jonatan Quezada <jonatan.quezada@chemeketa.edu>; Fernando Fernandez Lopez <fferna12@chemeketa.edu><br>
<b>Subject:</b> Re: [cisco-voip] refining dial peers for Fax<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">It really does seem to help from what I've seen. Users have accepted faxing is going to be slow. Most machines I find have it set to 9600 anyways.<o:p></o:p></p>
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<p class="MsoNormal">On Tue, May 8, 2018 at 2:17 PM, Anthony Holloway <<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">/squints eyes <o:p></o:p></p>
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<p class="MsoNormal">Not sure if sarcasm, or helpful.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Tue, May 8, 2018 at 12:48 PM Brian Meade <<a href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">I like to limit down to 9600. That seems to work out much better.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Tue, May 8, 2018 at 12:54 PM, Anthony Holloway <<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">What's the explanation for setting the fax machine itself to 14400, or the dial-peer for that matter, when for the most part SG3 is not supported, and SG3 gets spoofed down to G3, and the command "fax rate voice", which is the default,
already caps at 14400? <o:p></o:p></p>
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<p class="MsoNormal">I might have explained that poorly, but basically, I see 14400 speeds all the time, and I don't change the fax rate command nor set the speed on the machine.
<o:p></o:p></p>
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<p class="MsoNormal">I'm not challenging what you're saying, just trying to understand it. Fax has been a pain for me, just like everyone else, so the more I know, the better I can deal with it.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">I do like to avoid unnecessary config when possible, but in this case, I just don't know if there is proven evidence, that you need to do these two things.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Tue, May 8, 2018 at 11:35 AM Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Set the TX/RX rate at 14400 kbps and turn ECM. I would do that at the machine level first, and/or the dial-peer level second.<o:p></o:p></p>
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<p class="MsoNormal">Sent from my iPhone<o:p></o:p></p>
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On May 8, 2018, at 12:17, Jonatan Quezada <<a href="mailto:jonatan.quezada@chemeketa.edu" target="_blank">jonatan.quezada@chemeketa.edu</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">we are finding that after our sip cutover, that our faxes are happiest signalling over a T1 connection that originally we were trying to get away from, however trouble shooting was terrible and we are moving past having all voice traffic
on the SIP trunk. <o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Currently we are signalling for voice( calls ) only on the trunk and fax traffic can come in and out via that T1.<o:p></o:p></p>
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<p class="MsoNormal">My question is , still every so often we are seeing fax drops and incomplete page transmissions.<o:p></o:p></p>
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<p class="MsoNormal">looking at the controller the interface is solid no slips and seems to negotiate the connections just fine. but again every so often there are drops or sending fails altogether.<o:p></o:p></p>
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<p class="MsoNormal">We are wanting to try limiting the transmission rates but on the ATA190 and 191s you cannot rate limit on the device. It sounds like this needs to done at the dial peer level. if so what is the best starting configuration for the dial peers
that will handle on ly fax and go out a certain gateway that has the T1 on it?<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><a href="https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/115742-fax-modem-call-flows-00.html" target="_blank">https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/115742-fax-modem-call-flows-00.html</a><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Im looking at this call flow<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><a href="https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/115742-fax-modem-call-flows-00.html#anc7" target="_blank"><span style="font-size:10.5pt;font-family:"Arial",sans-serif;color:#6F53BC;border:none windowtext 1.0pt;padding:0in;background:white;text-decoration:none">Telco
- PRI - GW - MGCP - CUCM - SIP - ATA187 - Fax/Modem</span></a> <o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">except the ATAs are 190 and 191s<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">If we go the dial peer route, since the DID are not contiguous I will need a dial peer for each one huh? <o:p></o:p></p>
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<p class="MsoNormal">-- <o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">For immediate assistance please reach out to Chemeketa IT Help Desk at
<a href="tel:(503)%20399-7899" target="_blank">5033997899</a></span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">-or-</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">Visit the help center from your employee dashboard found here:</span><o:p></o:p></p>
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<p class="MsoNormal"><b><span style="font-size:13.5pt;font-family:"Arial",sans-serif"><a href="https://dashboard.chemeketa.edu/helpcenter/default.aspx" target="_blank">https://dashboard.chemeketa.edu/helpcenter/default.aspx</a></span></b><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:13.5pt;font-family:"Arial",sans-serif">Johnny Q</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:12.0pt;font-family:"Arial",sans-serif">Voice Technology Analyst - TelNet</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:13.5pt;font-family:"Arial",sans-serif">Chemeketa Community College</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.5pt"><a href="mailto:Johnny.Q@chemeketa.edu" target="_blank">Johnny.Q@chemeketa.edu</a></span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Arial",sans-serif">Building 22 Room 131</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Arial",sans-serif">Work
<a href="tel:(503)%20399-5294" target="_blank">5033995294</a></span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Arial",sans-serif">Mobile
<a href="tel:(971)%20218-2110" target="_blank">9712182110</a></span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial",sans-serif">SIP
<a href="tel:(503)%20540-6686" target="_blank">5035406686</a></span><o:p></o:p></p>
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