[asterisk-users] change dialing process on live call

[Khalil Khamlichi]

Thanks for your response, this works but we cannot hardcode this in the
dialplan, we need this to be done from an external application connected
either via manager or stasis.


On Sun, Aug 19, 2018, 11:14 AM Doug Lytle <support at drdos.info> wrote:

> On 08/19/2018 05:57 AM, Khalil Khamlichi wrote:
>
> Is there a way to add another extension to a live dial, for example
>
> Dial(PJSIP/1000,,)
>
> and after 20 secondes change it to
>
> Dial(PJSIP/1000&PJSIP/1001,,)
>
>
> This is a simple one.
>
>     exten => s,1,Dial(SIP/1000,20)
>     exten => s,n,Dial(SIP/1000&SIP/1001,20)
>     exten => s,n,Hangup()
>
> The first dial will ring with a 20 second timeout and proceed to the next
> dial and ring both extensions for 20 seconds and finally hangup
>
> Doug
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> 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/20180819/9058912d/attachment.html>