[Asterisk-Users] Call Back On Busy?

Sherwood McGowan madprofzero at yahoo.com
Mon Sep 26 12:57:50 MST 2005


Not a bad idea, thank you for that. I'll look into it
 


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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of BJ Weschke
Sent: Monday, September 26, 2005 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Back On Busy?


 Is there a functional reason why you'd use MeetMe here? I think probably
the easiest way to accomplish this is to use an DeadAGI script which can be
invoked via the 'h' extension in the context that would then perform the
functionality you're looking for and if they get through it should just
bridge the original caller back in. 


On 9/26/05, Sherwood McGowan <madprofzero at yahoo.com> wrote: 

Anyone else out there have some thoughts? The customer wants to be able to
control what can be redialed on busy, such as no international. I'm having
my doubts as to whether or not this can be done. My idea seems like it would
work, but after the customer hangs up, wouldn't the context stop processing?

 
Thanks,
SKM


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From: asterisk-users-bounces at lists.digium.com [mailto:
asterisk-users-bounces at lists.digium.com] On Behalf Of Damon Estep
Sent: Monday, September 26, 2005 10:15 AM 
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call Back On Busy?

 


This may not apply to your situation, but many ATAs and SIP phones have this
feature built in to the device. 

 

We use Linksys/Sipura and auto redial and last call return work without any
special setup.

 


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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com ] On Behalf Of Sherwood
McGowan
Sent: Monday, September 26, 2005 7:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Subject: [Asterisk-Users] Call Back On Busy?

 

I know it's been touched on before, but no answers have been found to the
best of my knowledge. I'm using a SIP only setup, with a sip provider giving
PSTN and would like to see if anyone has an idea for creating redial busy
using ${DIALSTATUS} and possibly MeetMe? 

 

I figure something like this, but want to get feedback

 

1. Get callers last dialed number, if international number, do not allow.

2. Playback a stuttertone to caller

3. Disconnect caller

4. Ring intended party check dial status. If busy, wait 120 seconds and try
again (do this for a total of 15 minutes)

5. If it's picked up, playback an announcement to the party and put them in
a meetme conference

6. Ring the original caller and bridge them to the meetme conference. 

 


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