Displaying 20 results from an estimated 8000 matches similar to: "Asterisk re-invites and billing"
2005 Jul 25
0
CDR Accounting/Billing Advise
Hello all,
I wondering if you guys could provide me with some advise. We have
developed a simple management tool for a small site which is
architected in the following way:
Server A <==> Server B <==> Server C
Server A is an asterisk server with a TE410P which connects to the
PSTN as well as to a couple of SIP VoIP termination providers.
Server B is another asterisk server
2008 Mar 10
1
Local music on hold -- mohinterpret=passthrough assymetrical ?
Hi list,
I'm planning and testing a distributed asterisk deployment
throughout several sites; each will be connected to the PSTN
and all of them among themselves via IAX trunks. Phones
will be SIP.
I guess I already "solved" (worked-around, actually) asterisk's
codec negotiation limitations regarding local G.711 utilization vs.
remote G.729 while minimizing
2008 Jun 14
1
World Most Economical Predictive Dialer!
Hi Tilghman!
> Clearly, you missed the point. Since there is a FREE predictive dialer out
> there, and your product costs something, you are not the world's cheapest
> predictive dialer.
I respect your wording and the way you or other people think on the list about difference between cheapest and free predictive dialer.
Surely
2006 Jan 11
1
asterisk with an external predictive dialer
Does anyone have any experience using asterisk with an external predicitve dialer, like MediaTel?
Specifically:
The predictive dialer dials out over T1 circuits.
It connects to asterisk via amphenol cable from an fxs card in the dialer to asterisk with a tdm2406 fxo card.
In the analog world, the dialer dials out through the t1 circuit, and the fxs card is plugged into a 66 block so the
2004 Sep 16
0
Predictive Dialer, Web & Inbound Phone System
Currently I have a predictive dialer that is web-enabled, as well as a
superdialer mechanism. I was wondering what kind of success people have
had with their dialers (lessons learned, etc.) and how I need some
direction on how to seamlessly integrate an inbound system with the
predictive dialer.
My thoughts are that inbound phone calls will be routed to a
receptionist first, who will then
2013 Feb 15
0
Rudi Ahlers invites you to Freelancer.com
15 February 2013
Hi,
Rudi Ahlers is inviting you to join Freelancer.com
(http://www.freelancer.com/users/7032909.html?utm_campaign=new_freelancer_invite&utm_medium=email&utm_source=freelancer&utm_content=new_freelancer_invite)
Rudi Ahlers
2006 Dec 12
1
AGI problema
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Verdana">Hi all. I've written a AGI in C language.
2020 May 16
0
PJSIP does not stop sending invites after call is canceled
Endpoint sends an INVITE
Asterisk send an INVITE to the Carrier
Carrier is down, does not even sends ACK
PJSIP sends several INVITES
End point sends
<--- Received SIP request (397 bytes) from UDP XXXX::50187 --->
CANCEL sip:xxxxxxx at xxxxxxx SIP/2.0
Via: SIP/2.0/UDP xxxxxxx
:50187;branch=z9hG4bK-524287-1---fbad0437cf02653d;rport
Max-Forwards: 70
To: <sip:xxxxx at xxxxx>
From:
2007 Sep 28
0
Proper trunk to connect two systems.
Hello,
I am replacing an exisiting call center with a new asterisk based
solution. This will initially consist of to phone servers. The first
being the main PBX, and the second being a predictive dialer. The
dialer will have sip extensions for all the agents, while the main pbx
will hand pretty much everything else.
The two boxes will be right next two each other, and are currently
2007 Oct 08
1
Sine Dialer, GNU dialer, VICIDial and others slightly OT?
Hello All,
I have a requirement to setup a predictive dialer for a customers call center.
I am asking for pros and cons of the different dialers available for
Asterisk. If you are going to send marketing material send it to my e-mail
directly please and not to the list. I was hoping to get the opinions of any
one using any of these dialers and what they liked and didn't like, ease of
2006 Mar 13
1
Need help implementing call center featuresofAsterisk
It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt
Florell
Sent: Monday, March 13, 2006 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2005 Jan 21
0
Help DIALSTATUS gives ANSWER when line is BUSY?
I'm running Asterisk CVS-v1-0-12/20/04.
I'm using PHP with Manager API Here is the code:
####################################################################
# Make call
####################################################################
$socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout);
if (!$socket) {
echo "$errstr ($errno)<br /\n";
} else {
2008 Oct 16
0
[Announcement] Billing Module for your SaaS web application is here: ServiceMerchant
[Announcement] Billing Module for your SaaS web application is here:
ServiceMerchant
== In two sentences
ServiceMerchant is open-source library, developed in Ruby, which takes
care of recurring billings and subscriptions of your SaaS
application.
The library sits on top of well known library ActiveMerchant and
therefore gives you the variety of choice among payment providers.
== In more details
2009 May 18
7
callcenter / dialer / predictive dialer / vicidial program is now open
This is a global message to all to announce our callcenter / dialer /
predictive dialer / vicidial program is now open.
Codecs: G711, GSM, G729, G723
Protocols: SIP
Duration Rate : 30/6 (6/6 with monthly minutes over 100,000)
Channels : 100 to start with , more on demand.
We are predictive dialer friendly , your account will not be shut off.
Contact us to do a test run.
Mike
2003 Jul 18
8
questions
Does anybody developed Predictive Dialer using Asterisk/Digium PBX?
Another question: does anybody developed an Dialer using the X100P board?
Julio
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2014 Feb 14
2
Dialer software for Asterisk...
I have a customer with a more or less unique need. Right now we
are using Wombat as a dialer software so they can contact clients for QA
purposes. Everything is working very well and their contact center
productivity is way up from the old manual dialing method.
The only thing we are having a problem with is that they have up to
5 phone numbers to contact a single customer. Obviously
2010 Jun 28
1
Never seen Problem !!!
One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0.
Today, when they downloaded , the CDR from the carrier site for 26th June 2010
, they see 50% calls are NEVER dialed by Dialer but it appears in CDR.
Amazingly, all the call durations are of 29-30 secs.
When we checked the status of the same in Dialer, lead is present there but
its marked as NEW which means Dialer has ever dialed
2006 Nov 26
0
Dialout to Meetme Fails?
I'm trying to use Asterisk (v1.2.11) make a callback that dials both
legs of the call into a Meetme() room together, but I keep getting
"conf-invalid" messages.
I created a callfile (/var/spool/asterisk/outgoing/out.call) that
specifies a Local channel (extension) which contains a Dial() command to
the "dialer", and an extension which contains a Dial() command to the
2006 May 18
1
SIP re-invite and billing
I know this may sound like a stupid question but I will put on my flame
retardant suit and ask anyway.
Is there any way to use/allow SIP reinvite and still track the length of the
call?
I realize that the whole idea of reinvite is that it takes the proxy out of
the media path which, from what I understand also kills the proxy's ability
to track the start/end time of the call for billing
2007 Jun 14
4
Que on A2Billing
Hello All,
I got one quick question on A2Billing.
Specs: -
- A2Billing v1.3
- OS CentOS 4.5
- Asterisk 1.2
- Zaptel 1.2
Did the installation and everything is working as it suppose to...
Using the A2Billing documentation, I created the RateCard, SIP Trunks,
and SIP Customers. I was also able to login using XLite Dialer and was
able to call out to my SIP Trunk also.
Now how can I remove the