Hi,
Trying to bridge SIP calls to an ss7 switch, it appears that when a call is
answered, this event isn't passed through either way, forcing me to first
manually run "Answer", else a Dial with timeout expires and drops. So an
extensions.conf with the following
exten => _1234.,1,Answer
exten => _1234.,n,Dial(Zap/r1/${EXTEN:2})
exten => _1234.,n,Hangup
..works fine, but without the first line, the call is never answered. Same
thing in the ss7->sip context:
exten => _12.,1,Answer
exten => _12.,n,Dial(SIP/${EXTEN}@sipprovider.com)
exten => _12.,n,Hangup
I've confirmed this by specifying a dial timeout, which triggers termination
of "Dial" after the timeout, even after the call has been answered (on
either end of the call). Doing the same thing using for example SIP-SIP
calls work as expected.
Am I doing something stupid, or might there be another cause?
Running asterisk-head and libss7-head of 27 Oct.
Cheers,
Charl
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