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
--
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New to Asterisk? Join us for
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
Vonage has a business offering, but they aren't really structured to provide
business quality support. I wouldn't use them for a business.
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
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
In my (limited) experience with rxfax, it has issues with large faxes. I
soon gave up on rxfax and switched to hylafax (which works much better).
Check the wiki for installation instructions. (And hylafax will
correctly hangup when the fax has completed/failed/whatever.)
Wes Baehr
-Original
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
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
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
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
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
***
; 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
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
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
I'm sure it took several hours, but, hey, he only has to sell one to get
his money back (:
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 22 Dec 2004, Luke Catranis wrote:
How much time did you waste on that?
-Original Message-
From: [EMAIL
The place it has for me is that my work is hesitant to pay for additional
training if there's nothing to show for it, like a certification.
They can't tell if I go to a $2000 training course and just goof off,
there's no tangible goods that they can see. If I got them to shell out
$3000 for my
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:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Taylor
Sent: Sunday, August 22, 2004 9:24 AM
To: [EMAIL PROTECTED];
-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
Once you've been ordained, do you have to wear black robes and a white
collar while working on Asterisk? :-)
Steve
Brian West wrote:
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]
But wait, that's not all! I, too, have a laser printer!
If you send me $50, I'll fire you off a certificate too!
You can be a Certified Asterisk Certification Certificate Buyer!
Enough. It is what it is. Don't like it? Don't pay for it.
Think it's a joke? Sure, but it's the same sick joke
Hello Guys,
I think this is not bad (Certification) While is a real certification
like Cisco , Novell, etc.
how many of us have a cisco certification or even Micro-$hit.
In my point of vew this 3K buck are well spended if I want to have the
skills quick to put hand-on. and as per Brian comment ,
:[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
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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
[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
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
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
Wow! That\'s great! Our company could really benefit from this level
of analysis. Previously we were using Nortel Merridian, and everyone is
used to that level of reporting. Your report(s) are the closest I\'ve
seen in their ability to provide the necessary statistics to manage a
call center.
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
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
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
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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
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[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
[EMAIL PROTECTED] wrote:
Hi again
How do I change the message played on initial pickup for after hours ??
Thanks in advance
Regards Mick
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[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
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
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
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
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
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
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
AJ.
If it's okay, I'd like to offer my 2 cents.
For switch type - it really doesn't matter what you pick, as long as both
ends (yours and your telco's) are configured to be same. I don't think
anyone has ever noticed any advantages or disadvantages with one switch
type over the other. What's
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