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

Vicky vicky.r at gmail.com
Mon Nov 13 04:11:33 MST 2006


Put canreinvite=no in asterisk sip user
extension . Some providers do not support reinvites and hence you get
silence i guess .


On 13/11/06, Yuri Veremeyenko <yuri.veremeyenko at gmail.com> 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/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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