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.

Sent from my iPhone

On May 8, 2018, at 12:17, Jonatan Quezada 
<[email protected]<mailto:[email protected]>> wrote:

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.

Currently we are signalling for voice( calls ) only on the trunk and fax 
traffic can come in and out via that T1.

My question is , still every so often we are seeing fax drops and incomplete 
page transmissions.

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.

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?

https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/115742-fax-modem-call-flows-00.html

Im looking at this call flow

Telco - PRI - GW - MGCP - CUCM - SIP - ATA187 - 
Fax/Modem<https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/115742-fax-modem-call-flows-00.html#anc7>

except the ATAs are 190 and 191s

If we go the dial peer route, since the DID are not contiguous I will need a 
dial peer for each one huh?








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Johnny Q
Voice Technology Analyst - TelNet
Chemeketa Community College
[email protected]<mailto:[email protected]>
Building 22 Room 131
Work 5033995294
Mobile 9712182110
SIP 5035406686
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