hi giorgio...
when i said "ring the calling phone first" I mean using a .call file!
I think now you are doing, in your .call files, something like this:
Channel: Zap/2/3391818250 or mISDN/1/3391818250/s
.....
.....
.....
and the rest to send this channel to the "calling phone".
This way, you have to wait the dial-out channel to be answered before connect it to the "calling phone". The fact that it works with TDM400 is only due to the fact that analog lines don't support call progress, so the call appears "answered" as soon as the fxo channel starts dialing out. With digital lines, instead, the channel is answered only when the remote party "pickups" the handset, that is correct. You will find same beaviour if you dial a voip line.
But, if you make your .call to connect to the "calling phone" first, then the call will go out ONLY when the calling party pickups! Which is correct!
By the way, doing like you do, you could have a situation where the remote party (33918....) pickups an incoming call and hears a ringing tone waiting the "calling phone" to answer.
Don't know if I make it clear... anyway... you're italian i think... if you want we can talk more about that privately, and in italian!
Bye
picciuX
2006/5/4, Giorgio Incantalupo <[EMAIL PROTECTED]>:
Hi picciux,
maybe it could work, even if I don't know how to call a phone a channel
and create a bridge between them.
I'd prefer to use Asterisk inner features like the auto-dial out call
moving a .call file to /var/spool/asterisk/outgoing: this works for
analog line but not for ISDN. But only in this case.....when normally
calling from a phone using an ISDN line, everything works fine. It is
the .call mechanism which does not work....and I want to understand why,
but it seems nobody had this problem before. It is also true not many
people use BRI ISDN.
Btw, thanx again.
Giorgio Incantalupo
picciuX wrote:
> probably it's better to auto-dial the calling phone first, and then
> let the established channel go out to the recipient!
> So when the "calling phone" answers, the call will go out to the
> recipient.
> Hope this helps...
>
>
> 2006/5/4, Giorgio Incantalupo < [EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>>:
>
> Hi,
> I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a
> monoBRI
> using chan-mISDN from beronet site.
> It seems to work all right except for autodial calls, monoBRI ISDN
> channel behaves differently waiting for the caller to answer and then
> continue.
> Asterisk console says:
>
> analog:
>
> -- Attempting call on Zap/2/3391818250 for
> [EMAIL PROTECTED]:1 (Retry 1)
> > Channel Zap/2-1 was answered.
> -- Executing DeadAGI("Zap/2-1", "exten2.py|ticket=19") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/exten2.py
>
> ISDN:
>
> -- Attempting call on mISDN/1/3391818250/s for
> [EMAIL PROTECTED]:1 (Retry 1)
>
> *Asterisk stops here for the caller to answer then go on to show
> the rest:*
>
> > Channel mISDN/1-u8 was answered.
> -- Executing DeadAGI("mISDN/1-u8", "exten2.py|ticket=21") in
> new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/exten2.py
>
> Why this pause? This is a problem because with ISDN the calling party
> phone does not ring.
> Is there some parametere to set in misdn.conf??
>
> TIA
>
> Giorgio Incantalupo
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