<span class="postbody">Hi,
<br>
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.<br>
However, when asterisk dials a number, Gizmo first answers then tries bridging 2 channels. Right after answer Asterisk starts playing the reminder.<br>It obviously results in hearing silence when the call is bridged (you've picked the phone).
<br><br>
I use Asterisk cmd Dial like this :
<br></span><span class="postbody"></span><br>
exten => s,1,Dial(SIP/NUMBER,30,rA(announce))
<br><br><span class="postbody">
which should play file "announce" to the called party once they answer.
<br><br>I also tried
<br></span><span class="postbody"><br></span>
exten => s,1,Dial(SIP/NUMBER,30,rG(default^play^1))
<br><span class="postbody"><br>which separates caller and callee,for the same purpose.
<br>
<br>Here's the asterisk console:
<br></span><br>
-- Executing SetCallerID("SIP/sipphone-cbfb", "NAME <NUMBER>") in new stack
<br> -- Executing NoOp("SIP/sipphone-cbfb", "Dialing 011XXXXXXXXXXXX
to deliver file /usr/vt/result/200611135/test") in new
stack
<br>
-- Executing SetVar("SIP/sipphone-cbfb", "__MSG=/usr/vt/result/200611135/98_011380673805838") in new stack
<br> -- Executing Dial("SIP/sipphone-cbfb",
"SIP/011380673805838@sipphone|45|rA(/usr/vt/result/200611135/test)")
in new stack
<br>
-- Called 011380673805838@sipphone
<br>
-- SIP/sipphone-ebf3 answered SIP/sipphone-cbfb
<br>
-- Playing '/usr/vt/result/200611135/98_011380673805838' (language 'en')
<br>
-- Attempting native bridge of SIP/sipphone-cbfb and SIP/sipphone-ebf3
<br>
<br><br><span class="postbody">
</span><span class="postbody">
It looks like Asterisk is starting to execute context/play announcement after the dialed channel answers.
<br>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.
<br>
Is there a workaround or a way to make Asterisk play the message when the call is bridged?<br><br>I use Asterisk CVS-HEAD built on 28 Oct 2006.<br><br>Any advice is highly appreciated.<br><br>Yuri<br></span><span class="postbody">
<br>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?<br></span><span class="postbody"><br><br></span>