Where can I find such ip-lists, please?
Am 05.06.2012 18:40, schrieb Alejandro Imass:
We use complete regional blocks from Wizcraft and blocking at minimum
all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We
block almost anything that is not our actual customer market and screw
On 06-06-12 11:41, Thorsten Göllner wrote:
Where can I find such ip-lists, please?
http://www.ipdeny.com/
Regards,
Patrick
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
Yesterday a customer was attacked from the following IP addresses so
add them to your blacklist:
iptables -A INPUT -s 37.8.119.75 -j DROP
iptables -A INPUT -s 37.8.22.240 -j DROP
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
We use complete regional blocks from Wizcraft and blocking at minimum
all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We
block almost anything that is not our actual customer market and screw
the rest.
On Tue, Jun 5, 2012 at 12:14 PM, Carlos Chavez cur...@telecomabmex.com wrote:
On Sat, 7 Apr 2012, sean darcy wrote:
I'm trying to run asterisk as asterisk. Which is harder than I
thought.
10.3.0. When I put a callfile into /var/spool/asterisk/outgoing, I get
this warning:
Unable to set utime on /var/spool/asterisk/outgoing/callfile.call:
Operation not permitted
There was a flurry of Vonage is going to unlock SIP activity last
year; did anything productive ever come of it?
Are *you* using your Vonage lines directly into Asterisk?
In lieu of that, for a 4 line small business that doesn't need to pay
Vonage $150 a month, who? Broadvoice? Someone else?
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Another State Of The Punctuation Mark question -
Vonage
There was a flurry of Vonage is going to unlock SIP activity last
year; did anything productive ever come of it?
Are *you* using your Vonage lines directly
On Tue, Sep 11, 2007 at 08:56:53AM -0400, Jay R. Ashworth wrote:
There was a flurry of Vonage is going to unlock SIP activity last
year; did anything productive ever come of it?
Are *you* using your Vonage lines directly into Asterisk?
In lieu of that, for a 4 line small business that
On Tue, Sep 11, 2007 at 09:32:06AM -0700, Eric Chamberlain wrote:
For several years now, we've used VoicePulse Connect
http://connect.voicepulse.com/ for our Asterisk IAX and SIP trunks.
Ravi and KP are both technical guys and know Asterisk extremely well.
They'd better be good; their business
On Tue, 11 Sep 2007, Jeff Bachtel wrote:
Broadvoice can't handle multiple lines being billed to the same account
and using the same SIP credentials, which is probably not too large a
deal for a 4 line install, but would quickly become unmanageable for
anything larger.
So it is not easy
On Tue, Sep 11, 2007 at 03:53:56PM -0400, Alex Balashov wrote:
On Tue, 11 Sep 2007, Jeff Bachtel wrote:
Broadvoice can't handle multiple lines being billed to the same account
and using the same SIP credentials, which is probably not too large a
deal for a 4 line install, but would
This probably came up before, but I have a faxing question for everyone.
I have a simple extension setup to use rxfax to receive faxes sent to
asterisk. It is:
exten = s,1,Answer()
exten = s,n,AbsoluteTimeout(300)
exten =
Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: Friday, March 09, 2007 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Another Faxing Question
This probably came up before, but I have
Hi so with my setup of asterisk 1.4 and installing freepbx on it, I
have everything working fine now except one thing, the remote console
keeps crashing after a reload. Any ideas?
___
--Bandwidth and Colocation provided by Easynews.com --
Hi
all,
That's my last one
for a while (I hope).
How can I (if at all
possible) make the 501 turn on the speaker phone as soon as a digit is dialed
(if the handset is not lifted)?Sort of likewhat a normal
speakerphone does.
The reason I want this is I want the 501 digitmap to be taken
Hi Mike,
As far as I know, you need to at least start the dialing (ie New call,
speaker, etc) for the digitmap to even come into play.
The only settings that I am aware of that you can try to change are
dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf.
Kevin
Mike
On Sun, 7 May 2006, Tofik Suleymanov wrote:
Hello folks,
firstly, thank you for your useful and fast answers !
Is there anybody using D-Link SIP phones ?
Are D-Link SIP phones ok to install in production environment ?
Give your comments please.
Tofik Suleymanov
I'll pipe in on this one.
On 7 May 2006, at 16:16, Aaron Daniel wrote:
On Sun, 7 May 2006, Tofik Suleymanov wrote:
Hello folks,
firstly, thank you for your useful and fast answers !
Is there anybody using D-Link SIP phones ?
Are D-Link SIP phones ok to install in production environment ?
Give your comments please.
Fred Noris wrote:
Hi:
I upgraded SuSE to 10 and Asterisk to trunk and now
after deleting all modules and previously compiled
stuff and recompiling asterisk, zaptel, and libpri, I
get this failure of asterisk to start:
[pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]:
loader.c:726
Hi:
I upgraded SuSE to 10 and Asterisk to trunk and now
after deleting all modules and previously compiled
stuff and recompiling asterisk, zaptel, and libpri, I
get this failure of asterisk to start:
[pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]:
loader.c:726 __load_resource: new style
Any disadvantage to always setting nat=yes for all UAs just
in case they end up behind a NAT at some point?
Canreinvite=no is always set since a few of our features
require it (transfers, etc.)
What is the impact of qualify=yes for 250-500 UAs?
Sorry about the unrelated questions about cisco phones, but does anyone
know how to set the second line up as a speed dial in the config file?
Or is that specifically a per-user basis setting?
Aaron
___
--Bandwidth and Colocation provided by
Hello
all,
I have
been posting some questions about this problems that I cannot yet solve, but I
think I have a better diagostic, so maybe someone can give me a clue why it is
happenning.
I have
Asterisk + AMPconfigured as a PBX with a Customer Center Queue with 4
agents that
hi
is this stuff still available?
roy
On 14. okt. 2004, at 16.10, Ben Merrills wrote:
I've been doing some work on a queue log analyser for a while now,
getting the basics in place, an example of which you can find at
the URL
below. However, just wondering what information people think
Title: Message
anyone got any ideas
on this?
TDM H323
Gateway SIP
Inbound H.323 call
'ip$200.93.237.82:12984/2853' detected.Channel OH323/R2853 created and
attached for inbound H.323 call 'ip$200.93.237.82:12984/2853'.Setting
channel 'OH323/R2853' (ip$200.93.237.82:12984/2853) native
My 7960 is configured for two lines, and I can turn the other appearance
buttons into speed dials from the menus, but is there any way to program
the speed dials in the SIPmacaddress.conf file?
/edg
___
Asterisk-Users mailing list
My 7960 is configured for two lines, and I can turn the other appearance
buttons into speed dials from the menus, but is there any way to program
the speed dials in the SIPmacaddress.conf file?
You can not: http://tinyurl.com/az4fp
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
Use the Directory or Services to create a speed dial list.
On 5/22/05, Nabeel Jafferali [EMAIL PROTECTED] wrote:
My 7960 is configured for two lines, and I can turn the other appearance
buttons into speed dials from the menus, but is there any way to program
the speed dials in the
Haven't done this yet Art but I will try it today at the
office...Thanks Jonathan
On Mon, 28 Mar 2005 00:30:32 -0600, Tim Litwiller [EMAIL PROTECTED] wrote:
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL
I'm working on it - I only started a week ago - and then I didn't know I
wanted to do all these other things with it. * is adictive!
Art Zemon wrote:
Tim Litwiller wrote:
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011.,1,SetCallerID(4153574000)
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
I want asterisk to use my pots line for local calls and voipjet
Tim Litwiller wrote:
so where did you put these lines?
exten = _1NXXNXX,1,SetCallerID(4153574000)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011.,1,SetCallerID(4153574000)
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Tim,
I did not use those lines. If you set
On Wed, Mar 09, 2005 at 06:38:24PM +1100, Callum McGillivray wrote:
Hey all,
Hi, welcome to this list
My apologies if this sounds blindingly obvious, but am I correct in saying
that I can use Asterisk to connect two extensions and make calls between
them without needing an actual telephone
Hey all,
My apologies if this sounds blindingly obvious, but am I
correct in saying that I can use Asterisk to connect two extensions and make
calls between them without needing an actual telephone line at all ?
As I said, probably blindingly obvious but my techies
have gone home for
Callum McGillivray wrote:
Hey all,
My apologies if this sounds blindingly obvious, but am I correct in
saying that I can use Asterisk to connect two extensions and make
calls between them without needing an actual telephone line at all ?
As I said, probably blindingly obvious but my
Hello,
Sorry for reposting the message, but I'm not sure the first post went
through.
I'm trying to figure out how to get Asterisk to dial an extension when a
call comes from the outside and contains the extension already.
(Somebody wants to call a user of Asterisk with extension 111
***
; defining the voice menu for incoming calls:
[fhostaffmenu]
exten = s,1,Ringing ; Make them comfortable with
some seconds of ringback
exten = s,2,Answer ; Answer the line
You haven't actually given them any ringing, you need to add this:
[mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 26, 2005 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Another BroadVoice Problem
This is a firewall/NAT issue. The UDP packets on your inbound RTP stream
and being dropped somewhere along
I finally got my incoming and outgoing working but outgoing
I cannot hear the called person, but the called person can hear me.
On incoming everything works perfect.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Awesome now i'm a minister!
On Wed, 22 Dec 2004 12:45:51 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Alexander Lopez wrote:
Agreed, You have a strong point about the Monopoly aspect of the whole
thing. My .02 would be to have this be a Digium product. Heck, Mark DID
invent the
Alternate Certification
For those of you who can't (or won't) shell-out the $3000+ for the 5 day
certification class,
here's a quicker way AND IT'S HALF THE MONEY!
www.metrotel.net/asterisk.htm
Asterisk is a good product.
Some people need certification.
A mature product needs certified
How much time did you waste on that?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Sunday, August 22, 2004 10:24 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Sunday, August 22, 2004 10:24 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another Asterisk Certification
Alternate Certification
For those of you who can't
]
[mailto:[EMAIL PROTECTED] On Behalf Of James Taylor
Sent: Sunday, August 22, 2004 10:24 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another Asterisk Certification
Alternate Certification
For those of you who can't (or won't) shell-out
]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another Asterisk Certification
Alternate Certification
For those of you who can't (or won't) shell-out the $3000+ for the 5 day
certification class,
here's a quicker way AND IT'S HALF THE MONEY
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Taylor
Sent: Sunday, August 22, 2004 9:24 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another Asterisk
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Taylor
Sent: Sunday, August 22, 2004 9:24 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another Asterisk Certification
Alternate Certification
For those of you who can't (or won't
But wait, that's not all! I, too, have a laser printer!
If you send me $50, I'll fire you off a certificate too!
You can be a Certified Asterisk Certification Certificate Buyer!
Enough. It is what it is. Don't like it? Don't pay for it.
Think it's a joke? Sure, but it's the same sick joke
] On Behalf Of James Taylor
Sent: Sunday, August 22, 2004 9:24 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another Asterisk Certification
Alternate Certification
For those of you who can't (or won't) shell-out the $3000
:[EMAIL PROTECTED] On Behalf Of Voip
Business
Sent: Wednesday, December 22, 2004 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Another Asterisk Certification
Hello Guys,
I think this is not bad (Certification) While is a real certification
like
certify this
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
What started out as a good thing for the community has veared it ugly
head and will come back to bite us in the ass. I give my respect to the
two companies that decided to put themselves 'out there' and attempted
to bring 'real world' certifications of knowledge in an area that is
Man, this is sick! :-)))
Isn't there a law against unclearly-marked jokes (except on April 1st, of
course)? Some people could even take you seriously! :-))
Most relevant points in the web page:
Starting a telephone company or consulting business is easy.
We authorize you to perform all
I, being one of the original Microsoft Certified guys, back then you
sent them $150 and you got the certificate and some logos (think 1980's
certification.)
In 1996 I was told by the company I was working for the certification
was needed if I was to keep my current salary.
What I saw was morons
[EMAIL PROTECTED] wrote:
Man, this is sick! :-)))
Isn't there a law against unclearly-marked jokes (except on April 1st, of
course)? Some people could even take you seriously! :-))
I'm rolling on the floor here :-
Regards,
Telmo.
Then I guess you haven't seen this one:
] On Behalf Of Brian West
Sent: Wednesday, December 22, 2004 10:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Another Asterisk Certification
No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!!
This is a joke right? I has to be. :P
bkw
.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Wednesday, December 22, 2004 12:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Another Asterisk Certification
What started out as a good thing
Alexander Lopez wrote:
Agreed, You have a strong point about the Monopoly aspect of the whole
thing. My .02 would be to have this be a Digium product. Heck, Mark DID
invent the thing and HE holds the copyright to it. I have faith in Mark
and what he can do when he gets back from France.
When
West
Sent: Wednesday, December 22, 2004 12:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Another Asterisk Certification
What started out as a good thing for the community has veared it ugly
head and will come back to bite us in the ass. I
Well give oej and steve some time here ... the project sure couldn't hurt
from more enterprise funding... lets just hope some of that makes it way
back to the root of the project. Also I was quick to judge their intentions
and I shouldn't have been... so guys lets give them some support and see
.. and from a newbie no less :-)
I have configured my BT101, and hooked it up to my * box. All is well.
I have entered the following in externsions.conf, and this bit works:
exten = 613,1,Answer
exten = 613,2,Playback(demo-echotest)
exten = 613,3,Echo
exten = 613,4,Hangup
If I pick up the
On Mon, 2004-12-06 at 17:40, Alan Ingleby wrote:
exten = 1000,2,Dial(Zap/1:555-1234,20,tr)
Change this to exten = 1000,2,Dial(Zap/1/5551234,20,tr)
Oh, and what extension do I use to reference an incoming call on my
FXO port? exten = 1 ??
You want the s extension.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills
Sent: Thursday, October 14, 2004 9:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] (Another) Queue log analyser
I've been doing some work on a queue
---
Nexus Technical Manager
n|m Nexus Management Inc
Neutral Bay
Sydney
Message: 4
Date: Fri, 15 Oct 2004 09:33:26 +0100
From: Ben Merrills [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] (Another) Queue log analyser
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL
Internet
T: 0870 8040862
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wayne
Sheppard
Sent: 14 October 2004 19:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] (Another) Queue log analyser
Very nice work Ben, thanks
I've been doing some work on a queue log analyser for a while now,
getting the basics in place, an example of which you can find at the URL
below. However, just wondering what information people think is most
useful in a log analyser?
At present it includes the following features:
# Time periods
: [Asterisk-Users] (Another) Queue log analyser
I\'ve been doing some work on a queue log analyser for a while now,
getting the basics in place, an example of which you can find at the
URL
below. However, just wondering what information people think is most
useful in a log analyser
I would like the source too so I can re-write it in non-.NET. Probably C or
PHP.
Matthew
- Original Message -
From: Joe Dennick [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 14, 2004 11:57 AM
Subject: Re: [Asterisk-Users] (Another) Queue log analyser
Wow! That\'s
Very nice work Ben, thanks. Here are some additional thoughts -
One segmentation that might be useful would be to add outbound calling
activities as a either a separate column or even view.
On agent stats, it would be useful to see login/logout stamps, login
time, ready/not ready time (if this
Another n00b question..
Realizing they will be all the same ext.
What is the maximum qty of phones one TDM400P FXS module will support
Or what would be the max REN alowable on that module
Again assuming north american usage etc...
Thanks
John B.
It's nice to be able to define the list of asterisk modules we want to
load from the /etc/sysconfig/zaptel file rather than directly in
/etc/init.d/zaptel. I'm using nufone and don't require anything but the
ztdummy (is the rtc-based module better, anyone?), so that's what I've
put here.
Adam Hart wrote:
I've also added support for SIP via TCP and the ability to change the
SIP port
It complains every time you click OK in the Options page about Changing
SIP port requires restart, even if you never looked at the SIP page
(and don't even have any SIP networks configured).
Kevin P. Fleming wrote:
Adam Hart wrote:
I've also added support for SIP via TCP and the ability to change the
SIP port
It complains every time you click OK in the Options page about Changing
SIP port requires restart, even if you never looked at the SIP page
(and don't even have any SIP
Adam Hart wrote:
That's a 'feature' - fixed, new version up
I found another 'feature' :) Although I couldn't get it to happen a 2nd
time, I had rung 18005558355 (via like2fone.com's sip server) and was
listening to the news and looking through the options dialog box, got
through all the options
With all the talk of SRV support in Asterisk, I thought I'd add support
in Firefly so enjoy. Thanks to Olle for helping me with it, explaining
the wonderful world of SIP and SRV to me. There's also an option to
disable it (seems to take quite a few DNS lookups for SRV) - warning
Duane may hunt
Hi!
I am looking for a software that can work as h.323 - sip gateway other than
asterisk and free. Someone can help me?
Thanks.
Mireia
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Try Vovida's Vocal, i think it does it.
Mireia Munoz de jesus wrote:
Hi!
I am looking for a software that can work as h.323 - sip gateway other than
asterisk and free. Someone can help me?
Thanks.
Mireia
___
Asterisk-Users mailing list
[EMAIL
Hi all;
I think I have the capacity issues figured out. My next question is
whether I can use asterisk for a redundant solution so that if any
hardware failure occurs on the phone switch, a spare PBX can route the
new calls. I have not been able to find this in the docs, and IIRC, it
is
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] another
[EMAIL PROTECTED] wrote:
Hi again
How do I change the message played on initial pickup for after hours ??
Thanks in advance
Regards Mick
___
Asterisk-Users mailing list
[EMAIL PROTECTED
think the above is correct ??
Bit how do I specify the after hours config ???
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of matt
Sent: Thursday, 18 December 2003 2:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] another
[EMAIL
Hi again
How do I change the message played on initial pickup for after hours ??
Thanks in advance
Regards Mick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[EMAIL PROTECTED] wrote:
Hi again
How do I change the message played on initial pickup for after hours ??
Thanks in advance
Regards Mick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of matt
Sent: Thursday, 18 December 2003 2:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] another
[EMAIL PROTECTED] wrote:
Hi again
How do I change the message played on initial pickup
If anyone is interested, I've trimmed one of Allison's recordings down
to the single word 'welcome', for use as a generic first message when a
line is answered.
I've put it up at:
http://jhcloos.com/sounds/asterisk/welcome.gsm
and will submit it to bugs.digium.com as well.
-JimC
I have an interesting problem now. I use asterisk to connect
to both FWD and a sip provider here in sweden. suddenly, (i know
my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try
to make a call using this provider. FWD still works fine, and I can call directly
put the core file into gdb, backtrace it
and then we'll have some useful information:
# gdb asterisk corefile
and issue bt on gdb console
or run asterisk directly into gdb :
# gdb --args asterisk -vvvgc
play with it and when it seg faults, issue a 'bt'
command
matteo.
Il lun, 2003-12-01 alle
Thanks Jose/Tom for responding to my Newbie questions. its much clearer
now. anyhow on to the next [unrelated question] here's the use case:
i will need one machine that will answer incoming calls - store the
caller's number [caller ID] and then prompt the caller to answer a
question by using
Look into AGI, there a re some examples out there, but it's very much
doable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brez
Sent: Monday, November 03, 2003 11:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Another newbie question
Thanks
== Parsing '/etc/asterisk/adsi.conf':
Found -- Accepting call from '890003' to '185' on channel
27, span 1 -- Executing Answer("Zap/27-1", "") in new
stack -- Executing Record("Zap/27-1",
"soundexampless:mp3") in new stack -- Playing
'beep'WARNING[360468]: File translate.c, Line 128
On Tue, 2003-10-28 at 04:43, Alexandru Coseru wrote:
== Parsing '/etc/asterisk/adsi.conf': Found
-- Accepting call from '890003' to '185' on channel 27, span 1
-- Executing Answer(Zap/27-1, ) in new stack
-- Executing Record(Zap/27-1, soundexampless:mp3) in new stack
--
This was triggered by the lack of an mp3 encoder. Without a backtrace
there's no way to know it's fixed for sure, but if you cvs update it
should at least fail cleanly and if not please place a bug in the bug
tracker.
Mark
On Tue, 28 Oct 2003, Alexandru Coseru wrote:
== Parsing
Hi, Most embarrased newbie evere here again.
Possibly another daft question. I have the digium
starter kit lite, so I've got the single FXO and FXS lines
All is working well with local sip phones able to dial other phones,
conferencing, MOH (Thanks Asterisrk-users list!) along with the one
On 2003-06-27 at 14:24, Chip Mefford ([EMAIL PROTECTED]) wrote:
Is anyone actually using * as a primary phone system in
a small/medium sized business with more than a dozen
stations and a real receptionist who handles calls?
As impressed as I am with asterisk, and as happy as we are with it
On Sat, 2003-06-28 at 02:10, Jim Gottlieb wrote:
On 2003-06-27 at 14:24, Chip Mefford ([EMAIL PROTECTED]) wrote:
Is anyone actually using * as a primary phone system in
a small/medium sized business with more than a dozen
stations and a real receptionist who handles calls?
As impressed
I'm getting ready to give asterisk another shot
here. Didn't have a lotta luck last time, about 7-8
months back.
I have been scanning the list all this time though,
lurking.
A question that comes up from time to time, that I have
yet to see answered is;
Is anyone actually using * as a primary
PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Another PRI based question
DID's on Asterisk are seen as extensions.
Mark
On Sat, 7 Jun 2003 [EMAIL PROTECTED] wrote:
In speaking to the representative at Verizon, we came to the conclusion
that DID numbers were not the correct solution
You hit the nail on the head in saying the tariffs in this area are
whacked. A block of 20 DID numbers increases the cost of the PRI about
$220 a month and that is with the configuration of 12 inbound and 11
outbound. If you want the configuration otherwise or more DID numbers it
costs even
In speaking to the representative at Verizon, we came to the conclusion
that DID numbers were not the correct solution; however we were told by
Verizon that they could do something called assign individual numbers to
the PRI. What this would in effect do is give us an additional phone
number
Well I would have went with DIDs however it really increases the pricing
of their plan plus we then have to split the channels up as incoming and
outgoing. It gets pretty complicated. They already deliver 10 digits in
from what I understand. I will inquire from them whether or not I can set
On Sat, 2003-06-07 at 21:04, [EMAIL PROTECTED] wrote:
Well I would have went with DIDs however it really increases the pricing
of their plan plus we then have to split the channels up as incoming and
outgoing. It gets pretty complicated. They already deliver 10 digits in
from what I
1 - 100 of 103 matches
Mail list logo
