Hello,
I am developing an asterisk autodialer. I am looking for the following
information:
1. Detailed Configuration Documentation for Asterisk Autodialer
2. Volume Testing Strategy
3. Lessons Learnt from past Asterisk Autodialer configuration
4. What are the different asterisk
a dialer and want people to recommend one, i think that's the
question you should be asking if not.
Cheers
Geraint
2009/8/25 Sanjoy Rath sanjoy_r...@hotmail.com
Hello,
I am developing an asterisk autodialer. I am looking for the following
information:
1. Detailed Configuration Documentation
Thanks Steve for helpful reply.
Cheers,
SR.
Date: Tue, 25 Aug 2009 10:26:37 -0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Autodialer
On Tue, Aug 25, 2009 at 10:04 AM, Sanjoy Rath sanjoy_r...@hotmail.com wrote
there is to know about
automatic dialers. Nobody's going to do that.
Sanjoy Rath wrote:
Thanks Steve for helpful reply.
Cheers,
SR.
Date: Tue, 25 Aug 2009 10:26:37 -0400
From: stot...@first-notification.com
.
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Sanjoy Rath
Sent: Tuesday, August 25, 2009
10:48 AM
To: Asterisk-Users
Subject: Re: [asterisk-users]
Asterisk Autodialer
Alex,
You are right. My questions are probably wide
as the load isn’t too great. You could even do
30-35 calls to account for hangups, etc.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sanjoy Rath
Sent: Tuesday, August 25, 2009
11:23 AM
To: Asterisk-Users
Subject: Re: [asterisk
Thanks Miguel. Have your configured GNUDialer before?
Date: Tue, 25 Aug 2009 11:22:16 -0500
From: mmol...@millenium.com.co
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Autodialer
Sanjoy Rath escribió:
Anyways I checked VOIP-info.org the information
, but pbx_spool will take
care of that easily as long as the load isn’t too great. You could even do
30-35 calls to account for hangups, etc.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sanjoy Rath
Sent: Tuesday, August 25, 2009
11:23
I have installed Asterisk. I want friends to connect to my asterisk server from
their SIP Phones are talk to me. I have tried two ways 1.) Have the Asterisk
server run within the firewall, opened all the ports for that server in
firewall port forwarding, does not work (One way audio issue). I
Hello,
I am also planning to implement Video Conf. There is AppConference you could
evaluate.
Curious, Did you get any response from anyone on your question?
Thanks,
SR. To: asterisk-users@lists.digium.com From: [EMAIL PROTECTED] Date: Fri, 6
Jun 2008 00:24:24 +0200 Subject:
When I dial one extension to the other, I get the call go into a HOLD music
instead of rining the other extention. Both extensions are SIP Softphone.
Following is the Asterisks CLI commandline log
-- Executing [EMAIL PROTECTED]:1] Park(SIP/500-08276430, ) in new stack
-- Started
When I dial one extension to the other, I get the call go into a HOLD music
instead of rining the other extention. Both extensions are SIP Softphone.
Following is the Asterisks CLI commandline log
-- Executing [EMAIL PROTECTED]:1] Park(SIP/500-08276430, ) in new stack
-- Started
dialing (701) is not an extension or UA but
rather a call parking slot.Tim NelsonSystems/Network SupportRockbochs
Inc.(218)727-4332- Original Message -From: Sanjoy Rath [EMAIL
PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comSent: Wednesday
@lists.digium.com Subject: Re: [asterisk-users] Call go into a
HOLD music instead Sanjoy Rath wrote: I am not dialing ext 701 but 700
from 500 ext. Do not know why its going to 701. Check your
features.conf, 700 by default is the parking extensions. It will place calls
on 701 first
I have a asterisk server. Two SIP Soft XLites are connected to the server. I am
able to make
calls from one SIP Phones to the other SIP Phones and landlines successfully.
The SIP Soft Phone on th eother side can hear my voice but I cannot hear their
voice.
They can call my local cell phone as
I have an asterisk server. Two SIP Soft XLites are connected to the server. I
am able to make calls from one SIP Phones to the other SIP Phones and landlines
successfully. The SIP Soft Phone on th eother side can hear my voice but I
cannot hear their voice. They can call my local cell phone as
Hello,
When I click on User menu, I get loading screen status. It runs indefinitely
without showing me
the user list and the user admin menu.
Any thoughts ?
Thanks,
Sanjoy.
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