Hi, I’m playing with call files and i wonder is it able to setup a connection somewhat like Dial(SIP/0100,D(www${EXTEN})
I have a internet modem with a pots connection. I use a FXO to make a connection to asterisk. and cant use. SIP/0100/${EXTEN}
Is it possible to create a call file that starts at an extent with use of this FXO connection?
You cannot add Dial application parameters in that way. For the way to do all non-trivial dial operations with call files, see:
https://wiki.asterisk.org/wiki/display/AST/Using+Callfiles+and+Local+Channels
Note that chan_sip will not be in the next release of Asterisk, later this year. You should be aiming to replace it with chan_pjsip.
I worked it out. But the call does not detected that the call is still ringing and marked as UP.
so i cant make a good call with that.
I know pjsip is prefered but my grandstream FXO does not handle end of calls well with pjsip.
Even with a proper analogue card, analogue answer supervision is a problem as not all PSTN operators provide it. With your setup the SIP side has already signalled answer before the digits are sent.
The only thing you can do from the Asterisk side is to wait for silence.
However, have you checked that the gateway is not capable of automatically waiting for secondary dialtone,
That seems strange. Even with providers with weird requirements, that’s not where I’d expect problems. You need to report the problem as, if it is not known, chan_sip will disappear with no solution for the problem in chan_psip,
If it is reported as a problem you’ll need to state what “does not handle end of calls well” means and provide a SIP trace using “pjsip set logger on”.
The only thing I can think of that would cause problems with ending a call is if the Grandstream is sending the wrong address in its Contact headers, in which case there is a rewrite contact option in chan_pjsip, to work round that.
This could happen if the Grandstream is inside NAT, and not compensating for it, and Asterisk is not in that particular NAT.
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