[asterisk-users] auto dial out multiple destinations

[Gustavo Cordeiro]

  I have the same problem using analog trunks (FXO), without solution. Now 
we only use digital (E1) or IP trunks (SIP/IAX) for auto-dial out.

  See this page for more information:

  
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out#Tipsandhints

  If you get the solution, please let me know! =)


Sds,
Gustavo

>From: Vieri <rentorbuy at yahoo.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>To: asterisk-users at lists.digium.com
>Subject: [asterisk-users] auto dial out multiple destinations
>Date: Tue, 24 Apr 2007 03:32:33 -0700 (PDT)
>
>Hi,
>
>I am searching for the most effective solution for the
>following scenario:
>
>Our users can call into our IVR menu and dial a
>specific extension and immediately hang up. This event
>should simply trigger Asterisk to make multiple
>simultaneous calls through a group of zap channels
>(5-10 calls). When the called parties answer, Asterisk
>should simply play a message and hangup.
>
>So I was thinking that I could simply add this in
>extensions.conf:
>
>exten => 844844,1,Playback(multicall-activated)
>exten => 844844,2,agi(multicallagi.php)
>exten => 844844,3,Hangup
>
>Then the AGI script would simply create a call file
>for each destination number and the format would be
>something like this:
>
>Channel: Zap/g0/5555555
>MaxRetries: 2
>RetryTime: 10
>WaitTime: 5
>Application: Playback
>Data: soundfile
>
>However, Asterisk doesn't wait for the destination to
>pick the phone up, so the playback ends prematurely
>and  the channel is closed. It works only if I use
>"Channel: SIP/5555 " (ie. it waits until the SIP phone
>answers and then plays the soundfile).
>
>I tried using
>Context: mycontext
>Extension: s
>Priority: 1
>
>and the same thing happens: the context lines are
>"run" immediately and even if the destination is not
>on line. The only difference here is that I can make a
>long loop so it plays back several times so that if
>the called party picks the phone up, there's a chance
>that they will here the looped playback. But this
>isn't very effective.
>
>Has someone done a "better" approach?
>Does someone know why only ZAP channels seem to
>misbehave (they are immediately considered "answered"
>when they are not) ?
>
>Also, could the multicallagi.php script "cut free" a
>zap channel in case all channels are already in use in
>group g0? Basically, these would be emergency calls
>and I wouldn't want them to be hanging around a long
>time in the outgoing queue. Also if multicallagi.php
>frees a busy channel it could get busy again before
>the call file is placed in "outgoing". So is there a
>way so that Asterisk knows that the call files I'm
>putting in outgoing are "emergency calls"? I know I
>could dedicate another group of channels or a single
>channel for these calls but I don't have any
>available.
>
>Thanks
>
>Vieri
>
>
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