[asterisk-users] Dial : Executing context/priority after bridge?

[Yuri Veremeyenko]

Hi,
I am using Asterisk to set up a reminder-like system, with asterisk
auto-dialing a user via SIP and playing a reminder file when the user picks
the phone. I use Gizmo service for SIP and I'm able to call through it.
However, when asterisk dials a number, Gizmo first answers then tries
bridging 2 channels. Right after answer Asterisk starts playing the
reminder.
It obviously results in hearing silence when the call is bridged (you've
picked the phone).

I use Asterisk cmd Dial like this :

exten => s,1,Dial(SIP/NUMBER,30,rA(announce))

which should play file "announce" to the called party once they answer.

I also tried

exten => s,1,Dial(SIP/NUMBER,30,rG(default^play^1))

which separates caller and callee,for the same purpose.

Here's the asterisk console:

-- Executing SetCallerID("SIP/sipphone-cbfb", "NAME <NUMBER>") in new stack
    -- Executing NoOp("SIP/sipphone-cbfb", "Dialing 011XXXXXXXXXXXX to
deliver file /usr/vt/result/200611135/test") in new stack
    -- Executing SetVar("SIP/sipphone-cbfb",
"__MSG=/usr/vt/result/200611135/98_011380673805838") in new stack
    -- Executing Dial("SIP/sipphone-cbfb",
"SIP/011380673805838 at sipphone|45|rA(/usr/vt/result/200611135/test)")
in new stack
    -- Called 011380673805838 at sipphone
    -- SIP/sipphone-ebf3 answered SIP/sipphone-cbfb
    -- Playing '/usr/vt/result/200611135/98_011380673805838' (language 'en')

    -- Attempting native bridge of SIP/sipphone-cbfb and SIP/sipphone-ebf3


 It looks like Asterisk is starting to execute context/play announcement
after the dialed channel answers.
The problem is that Gizmo SIP first answers and then tries bridging (that's
where the actual call is taking place), so my announcement is played before
the call and when I pick up I just hear the silence.
Is there a workaround or a way to make Asterisk play the message when the
call is bridged?

I use Asterisk CVS-HEAD built on 28 Oct 2006.

Any advice is highly appreciated.

Yuri

PS. I tried this on my local server with a local SIP account, and the
"bridge" step was absent. So it worked. Is it then a Gizmo issue or a
standard SIP way?
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