Displaying 20 results from an estimated 50000 matches similar to: "Screen Pop & Remote Agents"
2004 Jan 09
12
USA dial plan
Hi,
Do the callers in USA dialling from USA Telco lines always have to
prefix the CITY/AREA code with "1" in order
To successfully make a call to other USA destinations?
----
I have not been to USA (yet) :)
Ta
SJ
2004 Jan 09
1
Screen Pop & Remote Agents = Telemarketing
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of empire
underground
Sent: Friday, January 09, 2004 1:32 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Screen Pop & Remote Agents
> can I put a .csv file in the sql DB and have it dial from there? and
will I be able to set a
> Dial Plan to
2003 Oct 05
2
Good W2K softphone
Hi
U can visit the http://iaxclient.sf.net for some opensource underdevelopment
softphones.
Take Care
Obaid Amin Syed
>From: Chris Albertson <chrisalbertson90278@yahoo.com>
>Reply-To: asterisk-users@lists.digium.com
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] Good W2K softphone
>Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT)
>
>
>I haven't
2003 Nov 25
4
How to demo * on a notebook
I want to be able to demo * on a notebook at a client's site. This means no FXO gateways; just 2 sip phones (like SNOM) and maybe a softphone (GnoPhone?). I already have RH9 running on my notebook.
I would like to have one SIP phone dial and go through IVR before making a choice and ringing the other phone extensions. Of course the notebook would have to be running Asterisk.
How can i setup
2003 Oct 22
6
Running Asterisk and NAT on the same box?
Has anyone tried installing * on a box with two eth interfaces which is
acting as a NAT box? I have only one IP at this point and I would like
to get * working without all of the NAT issues. My idea is to run * on
my gateway (which is also running the firewall and masquerade services).
All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the
NAT screen, and will connect to the *
2003 Nov 14
7
Your thoughts..
I need to get your thoughts on something.. :)
I am trying to create a system to process the CDR call logs for
department accounting..
I think there are two ways of doing it.. Either I can create an AGI that
will run on the "h" extension and will lookup the last entry that
matches the account code of the call that just ended in the MySQL CDR
and calculate the call cost immediately..
2003 Sep 11
2
Segmentation fault due to SIP registration NUMBER 2
I assume that from your previous post that you are using iconnect
Is your register line in the format:
Register => 18005551212:1234@213.137.73.178/18005551212
I've had good luck using the IP address vs. the fully qualified
hostname. Remember that the register line goes in the [general] section
of sip.conf. Also, are you using the latest CVS release of *?
-----Original Message-----
2007 Mar 05
6
A New Phone Service - www.virtualphoneline.com
Dear Asterisk Users Mailing List - Non-Commercial Discussion,
I joined VirtualPhoneLine.Com service and am really enjoying the use of it.
VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, Google Talk or an IP Phone.
Have a look at the http://www.virtualphoneline.com/faq and
2004 Jan 24
13
Has Nufone gone belly-up
Folks,
I've ordered a new account from Nufone last month. Transferred money to
Nufone through their paypal account. I had communication with Nufone sales
up until two weeks back. Since then there were no replies to my emails.
I am afraid with this kind of unresponsiveness how one would run a reliable
service with this company. Have no bad feeling with Jeremy as the author of
widely used h323
2003 Dec 01
8
VoiceGlo
Hi,
VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll
Take a loock on http://www.voiceglo.com/
The softphone is IAX :)
Best regards,
Chris HARIGA
Techselesta Inc.
http://www.techselesta.com/
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2004 Sep 14
4
Sending Caller ID info in MD/USA
All,
Having trouble getting answer from Verizon. I believe Asterisk will let me specify a name and number that is sent to the PSTN (Verizon) of outgoing calls. For instance, if I have a client, First Bank, and their toll free number is 888-555-1234, I could send that name and number. Verizon is telling me that they will forward the number I send them, but the name will be my company's
2004 Jun 17
4
Problems with PRI with T410 messages
Hi all,
I have a box running asterisk with T410 connected to a Nortel DMS 100 switch
and another box running SER with grandstream phones on it
So if there is a call from the pstn it goes from the Nortel to the asterisk
and then to the SER box and finally to the phones.if the phone is busy or
the number is invalid the * box will first send an ALERT message to the
Nortel and say the call is going on
2003 Aug 17
2
no incoming packets & Sound: Recording overrun
On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote:
> Hello, and thank you for registering at gnophone.com. Your login
> information is listed below:
>
> Username: miernik
> Password: *******
> IAX Phone Number: 17002916107
>
> Please login as soon as possible to
> http://x.linux-support.net/directory/ to complete the
2003 Sep 11
1
Segmentation fault due to SIP registration N UMBER 2
Hello,
Don't know if this is related but I just got a segmentation fault today
while trying to register my new SNOM200 phone:
*CLI>
*CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request):
Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14'
NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request):
Registration from
2003 Sep 29
14
Help with GPL license of Asterisk
I would appreciate some help with this. I read the GPL license and basically it says you can do whatever you want with the software (sell, modify) as long as you include the source code, the License and make any changes you make available in the same manner to all others.
My questions is this: If I develop an external application (say a Call Center application or a GUI management application)
2003 May 14
20
Call forwarding
Yo,
Inspired by the example in the tips & tricks-section of
"http://www.junghanns.net/asterisk/", I built a more elaborate
call divert-feature. This one validates if the extension a call-forward
is to be set to is actually valid for the current context and
additionally saves this context into the DB and always uses it to
originate the divert from, as you can't expect the
2003 Nov 12
2
Media Negotiation Failed
Hi, I have this scenario
Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip:
64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than
* )
When a calls comes in Cisco 5300, this send this calls with SIP to *,
asterisk plays a welcome message and resend call to Cisco 3600 that have
4 analog lines connected... but after cisco play welcome message and
when
2006 Nov 15
7
Do Not Call List
The US has a Do Not Call list to which people can subscribe to prevent
being called by advertisers. Federal laws (strengthened by some state
and more local laws) assign penalties for calling people/phones on the
DNCL. Is there a query gateway that Asterisk (or an app using Asterisk)
can filter through to ensure a number is OK to call (not on the list)
before calling it?
--
(C) Matthew Rubenstein
2005 Apr 09
3
CallerID name lookup AGI script
Hi all,
My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote
an AGI script that does the following:
1) If it's a toll free number (800|888|877|866), set the CallerID name to
"TollFree Caller"
2) Use curl to look up the number in Google phonebook
3) If a business listing, set the CallerID name to business name, as is.
4) If it's a residential
2004 May 29
4
PlayTones problem
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi!
I am having problems with the PlayTones application and VoIP softphones.
I have the following in my extensions.conf:
exten => 123,1,Answer
exten => 123,2,PlayTones(Busy)
exten => 123,3,Hangup
But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call
just hangs up immediately.
I get the following on the console:
--