[asterisk-users] Answering machine detection for a second leg callgenerated by a call file.

Asterisk Man theasteriskman at gmail.com
Thu Mar 17 23:52:54 CDT 2011


This seems better. I will give it a try.
Thanks federico.

On Thu, Mar 17, 2011 at 11:10 PM, federico cabiddu <
federico.cabiddu at gmail.com> wrote:

> AMD is used mainly in scenarios like yours where an agent (the SIP
> extension) is called, then an outbound call is generated and finally
> the two legs are bridged. In your case you could call the Dial cmd
> using the M option. The argument of M can be a macro like this simple
> one:
>
> exten => s,1,Background(short_silence)
> exten => s,n,AMD()
> exten => s,n,GotoIf($[${AMDSTATUS}=MACHINE]?mach:humn)
> exten => s,n(humn),MacroExit
> exten => s,n(mach),Set(MACRO_RESULT=CONTINUE)
>
> So if an human is detected the legs will be bridged, if not the called
> party will be hangup and the next number will be called.
> The problem is, like previously said, the accuracy of the detection...
>
> Best regards,
>
> Federico
>
> 2011/3/17 Asterisk Man <theasteriskman at gmail.com>:
> > Thanks buddy,
> > But I think, AMD helps when I call customer first and then SIP extension.
> > Any other suggestion!
> >
> > On Thu, Mar 17, 2011 at 6:44 PM, Danny Nicholas <danny at debsinc.com>
> wrote:
> >>
> >> ________________________________
> >>
> >> From: asterisk-users-bounces at lists.digium.com
> >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Asterisk
> Man
> >> Sent: Thursday, March 17, 2011 8:13 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: [asterisk-users] Answering machine detection for a second leg
> >> callgenerated by a call file.
> >>
> >>
> >>
> >> Hi Group,
> >>
> >> I have following case scenario.
> >>
> >> Through call file, Asterisk makes a call to  SIP extension. When
> Extension
> >> answers the call, Asterisk reads customer numbers (set in callfile) and
> >> calls them one by one untill one of the customers answeres the call.
> Here
> >> customer and SIP extension gets patched and talk to each other.
> >>
> >> Now if outgoing call is answered by Answering machine,I don't want
> >> asterisk to patch it up with SIP extension. Please suggest me how this
> can
> >> be achieved.
> >>
> >> Thanking you in advance.
> >> --AM
> >>
> >>
> >>
> >> May or may not help – google for “Asterisk AMD”
> >>
> >> --
> >> _____________________________________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>               http://www.asterisk.org/hello
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >               http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110318/9f0885cb/attachment.htm>


More information about the asterisk-users mailing list