This seems better. I will give it a try.<br>Thanks federico.<br><br><div class="gmail_quote">On Thu, Mar 17, 2011 at 11:10 PM, federico cabiddu <span dir="ltr"><<a href="mailto:federico.cabiddu@gmail.com">federico.cabiddu@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">AMD is used mainly in scenarios like yours where an agent (the SIP<br>
extension) is called, then an outbound call is generated and finally<br>
the two legs are bridged. In your case you could call the Dial cmd<br>
using the M option. The argument of M can be a macro like this simple<br>
one:<br>
<br>
exten => s,1,Background(short_silence)<br>
exten => s,n,AMD()<br>
exten => s,n,GotoIf($[${AMDSTATUS}=MACHINE]?mach:humn)<br>
exten => s,n(humn),MacroExit<br>
exten => s,n(mach),Set(MACRO_RESULT=CONTINUE)<br>
<br>
So if an human is detected the legs will be bridged, if not the called<br>
party will be hangup and the next number will be called.<br>
The problem is, like previously said, the accuracy of the detection...<br>
<br>
Best regards,<br>
<div class="im"><br>
Federico<br>
<br>
2011/3/17 Asterisk Man <<a href="mailto:theasteriskman@gmail.com">theasteriskman@gmail.com</a>>:<br>
</div><div><div></div><div class="h5">> Thanks buddy,<br>
> But I think, AMD helps when I call customer first and then SIP extension.<br>
> Any other suggestion!<br>
><br>
> On Thu, Mar 17, 2011 at 6:44 PM, Danny Nicholas <<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>> wrote:<br>
>><br>
>> ________________________________<br>
>><br>
>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Asterisk Man<br>
>> Sent: Thursday, March 17, 2011 8:13 AM<br>
>> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
>> Subject: [asterisk-users] Answering machine detection for a second leg<br>
>> callgenerated by a call file.<br>
>><br>
>><br>
>><br>
>> Hi Group,<br>
>><br>
>> I have following case scenario.<br>
>><br>
>> Through call file, Asterisk makes a call to SIP extension. When Extension<br>
>> answers the call, Asterisk reads customer numbers (set in callfile) and<br>
>> calls them one by one untill one of the customers answeres the call. Here<br>
>> customer and SIP extension gets patched and talk to each other.<br>
>><br>
>> Now if outgoing call is answered by Answering machine,I don't want<br>
>> asterisk to patch it up with SIP extension. Please suggest me how this can<br>
>> be achieved.<br>
>><br>
>> Thanking you in advance.<br>
>> --AM<br>
>><br>
>><br>
>><br>
>> May or may not help – google for “Asterisk AMD”<br>
>><br>
>> --<br>
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