just fine in both cases, but then as soon as asterisk bridges the two channels, the remote end sends a "Call/Leg Transaction Doesn't Exist" error and hangs up the line.
Apparently it doesn't like our reinvites for some reason. Any SIP experts out there able to figure out why the ATA doesn't like them? the Pingtel seems to be okay with them.
I set "canreinvite=no" in the sip.conf for both instruments. That doesn't seem to have made any difference.
Does anyone know of a fix for this?
I have talked ATA<->ATA when they're both on different networks; that's the oddest part. . .
B.
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