On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > Just a guess (without knowing about your network), but are the two ends > points on public networks and visible to one another? If not the reinvite > may be passing an internal (nat'ed) address to the other and the connection > will fail...just a though
t38modem -tt -o /var/log/t38modem.log --no-h323 -u 91 --sip-listen udp\$127.0.0.1:6060 --ptty +/dev/ttyT380,+/dev/ttyT381 --route 'modem:.*=sip:<dn>@127.0.0.1:5061' --route 'sip:.*=modem:<dn>' --sip-register [email protected]:5061,password I tried it with a global IP (instead of 127.0.0.1) - same behavior. The point is, that the receiving part, which initiates the t.38 switch, doesn't sent the switch to the ISP. It is blocked / ignored by asterisk at all - don't know why it isn't sent to the ISP. The extension is a normal pjsip extension with these additional options: t38_udptl : true t38_udptl_ec : redundancy t38_udptl_ipv6 : false t38_udptl_maxdatagram : 400 t38_udptl_nat : no (or yes - doesn't matter) The trunk looks exactly the same: t38_udptl : true t38_udptl_ec : redundancy t38_udptl_ipv6 : false t38_udptl_maxdatagram : 400 t38_udptl_nat : no (or yes - doesn't matter) Thanks, Michael -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
