[asterisk-ss7] libss7 reporting answer event to asterisk

[Matthew Fredrickson]

On Oct 29, 2006, at 3:14 AM, Charl Barnard wrote:

> 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?

No, it is supposed to work like this.  If you have configured a PRI 
before in asterisk, you would have seen similar behavior.  The reason 
for this is so that you can play early media on the channel before you 
start billing the call (before it is Answer()'d).  That is strange that 
it is not answered at all though without that.  Are you sure that the 
SIP endpoint you are calling actually answers the call in the 
signalling?

Matthew Fredrickson