Put canreinvite=no in asterisk sip user extension . Some providers do not support reinvites and hence you get silence i guess . <br><br><div><span class="gmail_quote">On 13/11/06, <b class="gmail_sendername">Yuri Veremeyenko
</b> <<a href="mailto:yuri.veremeyenko@gmail.com">yuri.veremeyenko@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
<span>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><br> exten => s,1,Dial(SIP/NUMBER,30,rA(announce)) <br><br><span> which should play file "announce" to the called party once they answer. <br><br>I also tried
<br></span><span><br></span> exten => s,1,Dial(SIP/NUMBER,30,rG(default^play^1)) <br><span><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> 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><span class="sg"><br>Yuri<br></span></span><span><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?
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