[Freeswitch-users] tcp call misses sip message

[RobertT]

Well, I start 2 user agents. Each of them successfully registers as 1000 &
1001 extensions via tcp SIP transport. Then I issue a call, say from 1000 to
1001, and watch it being disconnected in several seconds by recieving client
due to abovementioned conditions (no completing answer from FS). Why is it
happening???
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