On 13/11/06, Yuri Veremeyenko
<[EMAIL PROTECTED]> wrote:
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/[EMAIL PROTECTED]|45|rA(/usr/vt/result/200611135/test)") in new stack
-- Called [EMAIL PROTECTED]
-- 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?
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
