[asterisk-users] Outgoing SIP calls dropped after 30 seconds.

[Grey Man]

To get to the bottom of it I'd recommend determining why the ACKs are
not getting through to Asterisk rather than trying to work around it.
I'm actually suprised Asterisk terminates the call by default when it
doesn't get the ACK to it's 200 Ok response that must be new for
1.4.22 as I haven't seen that behaviour in earlier versions. In my
opinion it's unwarranted behaviour, if Asterisk is getting RTP then it
should leave the call up irrespective of whether it gets an ACK or
not.