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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">jonatan.quezada@chemeketa.edu</a>> 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
SIP trunk.
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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">https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/115742-fax-modem-call-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)">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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<div><font face="arial, helvetica, sans-serif" size="2">For immediate assistance please reach out to Chemeketa IT Help Desk at 5033997899</font></div>
<div><font face="arial, helvetica, sans-serif" size="2">-or-</font></div>
<div><font face="arial, helvetica, sans-serif" size="2">Visit the help center from your employee dashboard found here:</font></div>
<div><font face="arial, helvetica, sans-serif" size="4"><b><a href="https://dashboard.chemeketa.edu/helpcenter/default.aspx" target="_blank">https://dashboard.chemeketa.edu/helpcenter/default.aspx</a></b><br>
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<div dir="ltr"><font face="arial, helvetica, sans-serif" size="4">Johnny Q</font></div>
<div dir="ltr"><span style="font-family:arial,helvetica,sans-serif;font-size:small">Voice Technology Analyst - TelNet</span><font face="arial, helvetica, sans-serif" size="4"><br>
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<div><font face="arial, helvetica, sans-serif" size="4">Chemeketa Community College</font></div>
<div><span style="font-size:12.8px"><a href="mailto:Johnny.Q@chemeketa.edu" target="_blank">Johnny.Q@chemeketa.edu</a></span></div>
<div><font size="1"><span style="font-family:arial,helvetica,sans-serif">Building 22 Room 131</span><br>
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<div><font face="arial, helvetica, sans-serif" size="1">Work 5033995294</font></div>
<div><font face="arial, helvetica, sans-serif" size="1">Mobile 9712182110</font></div>
<div><span style="font-family:arial,helvetica,sans-serif;font-size:x-small">SIP 5035406686</span><br>
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