I like to limit down to 9600. That seems to work out much better. On Tue, May 8, 2018 at 12:54 PM, Anthony Holloway < [email protected]> wrote:
> 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? > > 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. > > 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. > > 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. > > On Tue, May 8, 2018 at 11:35 AM Ryan Huff <[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]> >> 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 <(503)%20399-7899> >> -or- >> Visit the help center from your employee dashboard found here: >> *https://dashboard.chemeketa.edu/helpcenter/default.aspx >> <https://dashboard.chemeketa.edu/helpcenter/default.aspx>* >> >> >> Johnny Q >> Voice Technology Analyst - TelNet >> Chemeketa Community College >> [email protected] >> Building 22 Room 131 >> Work 5033995294 <(503)%20399-5294> >> Mobile 9712182110 <(971)%20218-2110> >> SIP 5035406686 <(503)%20540-6686> >> >> _______________________________________________ >> cisco-voip mailing list >> [email protected] >> https://puck.nether.net/mailman/listinfo/cisco-voip >> >> _______________________________________________ >> cisco-voip mailing list >> [email protected] >> https://puck.nether.net/mailman/listinfo/cisco-voip >> > > _______________________________________________ > cisco-voip mailing list > [email protected] > https://puck.nether.net/mailman/listinfo/cisco-voip > >
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