Yes, ECM off. I’d also take a look at the user behaviors; in today’s world users tend to expect better performance from faxing than the faxing technology was ever designed or intended for (Ex. Sending a single 100+ page fax).
Faxing technology is less and less used (I know, still used heavily in some verticals); as such, it is less often a priority for vendors that make devices and firmware and can be buggy, especially when inter operating protocols and transport mediums. T.38 is a good option over sip because it is a packetized transmission verses analog modulation. Keep in mind, faxing is a real time transmission; meaning that the sending and receiving and both have to be active participants in the transmission at the same time (even though the machines on either end of the transmission may have memory buffering technology). As such, any interruption to that transmission can cause service failure. Faxing over sip can also have some service failure due to UDP transmission over TCP transmission, especially in the case of poor QoS or high jitter (because TCP will re-transmit some packet loss that may help save a transmission). Sent from my iPhone On May 8, 2018, at 12:37, Jonatan Quezada <[email protected]<mailto:[email protected]>> wrote: turn off ECM right? and I feel like this configuration works best for passthrough correct? when would one go to T38? can it go T38 with this type of 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 our ATA s are 190 and 191s On Tue, May 8, 2018 at 9:35 AM, Ryan Huff <[email protected]<mailto:[email protected]>> wrote: 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? -- For immediate assistance please reach out to Chemeketa IT Help Desk at 5033997899 -or- Visit the help center from your employee dashboard found here: https://dashboard.chemeketa.edu/helpcenter/default.aspx Johnny Q Voice Technology Analyst - TelNet Chemeketa Community College [email protected]<mailto:[email protected]> Building 22 Room 131 Work 5033995294 Mobile 9712182110 SIP 5035406686 _______________________________________________ cisco-voip mailing list [email protected]<mailto:[email protected]> https://puck.nether.net/mailman/listinfo/cisco-voip -- For immediate assistance please reach out to Chemeketa IT Help Desk at 5033997899 -or- Visit the help center from your employee dashboard found here: https://dashboard.chemeketa.edu/helpcenter/default.aspx Johnny Q Voice Technology Analyst - TelNet Chemeketa Community College [email protected]<mailto:[email protected]> Building 22 Room 131 Work 5033995294 Mobile 9712182110 SIP 5035406686
_______________________________________________ cisco-voip mailing list [email protected] https://puck.nether.net/mailman/listinfo/cisco-voip
