<div dir="ltr">I like to limit down to 9600. That seems to work out much better.</div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, May 8, 2018 at 12:54 PM, Anthony Holloway <span dir="ltr"><<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">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?<div><br></div><div>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.<div><div><br></div><div>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.</div><div><br></div><div>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.</div><div><div class="h5"><div><br><div class="gmail_quote"><div dir="ltr">On Tue, May 8, 2018 at 11:35 AM Ryan Huff <<a href="mailto:ryanhuff@outlook.com" target="_blank">ryanhuff@outlook.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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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.<br>
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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><wbr>> wrote:<br>
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<div dir="ltr">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
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<div>Currently we are signalling for voice( calls ) only on the trunk and fax traffic can come in and out via that T1.</div>
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<div>My question is , still every so often we are seeing fax drops and incomplete page transmissions.</div>
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<div>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.</div>
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<div>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?</div>
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<div><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/<wbr>support/docs/voice-unified-<wbr>communications/unified-border-<wbr>element/115742-fax-modem-call-<wbr>flows-00.html</a><br>
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<div>Im looking at this call flow</div>
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<div><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" style="margin:0px;padding:0px;border:0px;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-variant-numeric:inherit;font-variant-east-asian:inherit;font-weight:400;font-stretch:inherit;font-size:14px;line-height:inherit;font-family:CiscoSans,Arial,sans-serif;vertical-align:baseline;color:rgb(111,83,188);text-decoration:none;word-wrap:break-word;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255)" target="_blank">Telco
- PRI - GW - MGCP - CUCM - SIP - ATA187 - Fax/Modem</a> <br>
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<div>except the ATAs are 190 and 191s</div>
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<div>If we go the dial peer route, since the DID are not contiguous I will need a dial peer for each one huh? </div>
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