doesn't send the 1 and you will see the 1 is not
there.
On Fri, Dec 3, 2010 at 8:27 AM, John Novack
jnov...@stromberg-carlson.org wrote:
And yet SOME providers SEND the 1
Abiding by some standard would be nice!
John Novack
C F wrote:
When sending CLID in the US it should never contain more
When sending CLID in the US it should never contain more than 10
digits (don't include the 1). In fact some providers will BLOCK your
call if you do.
On Thu, Dec 2, 2010 at 2:24 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Some discussion on other lists regarding this, but the + should
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote:
Sorry,
what I meant was:
server*CLI remove extension (hit tab)
segfault..
1.4.22
It could be an extension name Where is the error trapping if this is the
case.. Who writes this shit?
If you get hurt do you blame your
r...@pbx:~# uptime
23:10:15 up 606 days, 9:38, 1 user, load average: 0.31, 0.08, 0.02
Customer called they are having a scheduled power outage for most of
the day because of construction if I can shut down the machine
gracefully. So I decided to run uptime first.
Enjoy
--
Search the archives you will get your answer.
On Mon, Nov 15, 2010 at 1:35 PM, Cassius Smith cass...@cassius.org wrote:
Hi all,
I have had (what I consider) an odd request. The installation I'm working on
now is an office on a multi-floor building. They 're looking for some kind
of solution
Wow you actually wrote code that could be accomplished for less than
$20.00 with sandman.
On Mon, Nov 15, 2010 at 2:34 PM, Dr. Michael J. Chudobiak
m...@avtechpulse.com wrote:
On 11/15/2010 01:35 PM, Cassius Smith wrote:
Hi all,
I have had (what I consider) an odd request. The installation I'm
You see the problem is that asterisk will send as many packets as its
admin does on the list. There is no way to change that. I suggest you
first change the amount of packets per second you send.
On Thu, Nov 4, 2010 at 5:38 AM, ali anjum aliraza_an...@hotmail.com wrote:
Hi,
(I have install
insecure=very should fix it.
On Wed, Nov 3, 2010 at 4:08 AM, Gregory Malsack gmals...@gmellc.com wrote:
Can anyone tell me why my inbound calls keep getting rejected with 401?
Here’s the debug information:
--- SIP read from UDP:147.135.32.221:5060 ---
INVITE
Joel, after sending my previous posts I did realize your points might
have some validity - and hence I owe you an apology - and that is if
you are a telco or hosted pbx provider then strict fail2ban is not
that good of a solution. While I was talking strictly from a PBX
vendors point of view,
On Tue, Nov 2, 2010 at 11:16 AM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Tuesday, November 02, 2010 10:06 AM
To: Asterisk Users Mailing List - Non
Like I said before RUBBISH.
One should just ban/block IPs that are attacking you and not let them
connect at all. Not just protect against them with fancy passwords.
BTW, even your fancy passwords are breakable, can't wait for the day
that you'll wake up and smell the coffee.
On Sun, Oct 31, 2010
On Sun, Oct 31, 2010 at 12:45 PM, Joel Maslak jmas...@antelope.net wrote:
On Oct 31, 2010, at 9:57 AM, Jeff LaCoursiere j...@sunfone.com wrote:
This only tells you after it is way too late that you now have upstream
bills to wrangle with your carriers about, or (like in my case) that your
, C F shma...@gmail.com wrote:
Like I said before RUBBISH.
One should just ban/block IPs that are attacking you and not let them
connect at all. Not just protect against them with fancy passwords.
BTW, even your fancy passwords are breakable, can't wait for the day
that you'll wake up
You kidding?
On Sat, Oct 30, 2010 at 3:43 PM, Joel Maslak jmas...@antelope.net wrote:
Is there really any benefit to blocking these, if you use good passwords?
On Sat, Oct 30, 2010 at 1:20 PM, Warren Selby wcse...@selbytech.com wrote:
I'm experiencing this on one of my clients servers. The
the monitoring period. The monitoring period is usually one hour, but can be
anything (5, 60, or 1440 minutes in this case).
On Oct 30, 2010, at 6:53 PM, C F shma...@gmail.com wrote:
You kidding?
On Sat, Oct 30, 2010 at 3:43 PM, Joel Maslak jmas...@antelope.net wrote:
Is there really any benefit
The other way is to use an RC3 from vikingelectronics.com
http://www.vikingelectronics.com/products/view_product.php?pid=217
On Mon, Oct 18, 2010 at 7:53 PM, C F shma...@gmail.com wrote:
Ah Sandman http://sandman.com use a relay that goes onto an fxs port,
call that fxs port and you have
I have had where the Phone provider (this was a PRI) cut long distance
service to a box that was compromised till we called them to assure
them that the security holes where fixed.
On Fri, Sep 17, 2010 at 1:10 PM, Jeff Brower jbro...@signalogic.com wrote:
All-
Recently an Asterisk server we
Just put in:
Answer()
Wait(1.5)
On Fri, Sep 17, 2010 at 11:00 AM, Ujjval Karihaloo
ujj...@simplesignal.com wrote:
With some carriers the initial Audio (2-4 secs) seems to get cut off when
using a Auto Attendant or Conf Meetme.
Is there any known remedies for that. Just want to know if others
7, 2010 at 5:29 AM, Barry O'Donovan
barry+asterisk-us...@opensolutions.ie wrote:
On 07/09/10 04:15, C F wrote:
Dial with M option
I don't see how executing a macro BEFORE connecting to the called
channel will help with my issue. Am I missing something?
Thanks,
Barry
Wow thats a LONG signature.
Oh and BTW, the future of this world, how many years in the future?
On Tue, Sep 7, 2010 at 9:16 AM, Frenette, Rob
rob.frene...@bullyingcanada.ca wrote:
Hi,
Does Asterisk, provide the option within its voice mail system to be able
to do the following: We want to
Dial with M option
On Mon, Sep 6, 2010 at 10:02 AM, Barry O'Donovan
barry+asterisk-us...@opensolutions.ie wrote:
Hi folks,
After a fairly extensive Google trawl, I don't think the following is
possible but would appreciate confirmation from anyone else who has
tried something similar.
I
Last time I analyzed this (I believe back in 1.2) there was no way of
telling. However a blind transfered call would generate 2 CDR
recoreds:
1. For the part of the call with the transferrer and transfered.
2. For the part of the call with the transferee and transfered.
The call duration for the
If you post the same question 10 times you have more chances to get an
answer please repost and additional 8 times.
On Tue, Aug 24, 2010 at 4:34 AM, Janu Mukherjee janu.mu...@gmail.com wrote:
Hi All,
I have this requirement. I have an xlite client registered with asterisk
server. And from
BTW, I believe after all said everyone agrees that non of the
providers send onto the PSTN more than 10 digits the only question is
what they send to their customers.
On Wed, Aug 11, 2010 at 1:35 AM, John Novack
jnov...@stromberg-carlson.org wrote:
C F wrote:
The correct way in the US
The correct way in the US is to send NPANXX without the 1, in fact
ATT used to display unknown if 11 digit was received.
Vonage decided to add the 1 so that their fancy stupid boxes can dial
back directly from the box. Sorry for the rant but it has created me
more headaches that I care to
Is asterisk and the SIP device behind the same router?
Most routers will not redirect internal NAT requests. So that if you
are trying to have port forwarding done but the request and the
forwarding destination are on the same interface it won't work.
On 8/3/10, Nasir Javaid
In most cases wait(.5) will do. I would not recommend using
answer(2000) as that answers the channel, which means you start
getting billed.
On 8/2/10, Peder pe...@networkoblivion.com wrote:
I am using T1's and didn't think the spill would take that long.
PRI no, EM yes.
Some PRI take that
Isn't that trademarked? :P
On Tue, Aug 3, 2010 at 9:28 AM, Alan Lord (News) alansli...@gmail.com wrote:
http://gigaom.com/2010/08/03/2600hz-project/
--
The Open Learning Centre
http://www.theopenlearningcentre.com
--
_
On Tue, Jul 13, 2010 at 8:47 PM, C F shma...@gmail.com wrote:
I agree with horns you'll usually get better coverage. I have done
this in the past with 5 speakers for a 30k sq ft warehouse very good
coverage. Using bogen horns. This was for a 300ft by 100ft warehouse.
Starting at 30 ft
On Wed, Jul 14, 2010 at 5:03 AM, liuxin nyliuxin...@gmail.com wrote:
Hi,
probably a misconfiguration or you havent plugged the cable in yet.
OMG you are right, I forgot to plug in the cable. Hey but wait which
cable you talking about?
2010/7/14 C F shma...@gmail.com
It has nothing to do
that would be 33 speakers. I
think that's way too much of an overkill.
thanks,
Bruce
On Mon, Jul 12, 2010 at 1:05 AM, C F shma...@gmail.com wrote:
In my experience using height for radius works, for example if you
have a 20 ft high ceiling then the coverage for one speaker would be
40 ft diameter
not saying your wrong
I am just trying to understand why it happens.
On Mon, Jul 12, 2010 at 7:56 AM, C F shma...@gmail.com wrote:
I have found that sometimes shutting down the machine waiting a full
minute while the power cable is unplugged then restarting can fix such
problems if it's power related
I have found that sometimes shutting down the machine waiting a full
minute while the power cable is unplugged then restarting can fix such
problems if it's power related.
On Fri, Jul 9, 2010 at 12:42 PM, Jerry Geis ge...@pagestation.com wrote:
I have a TE205P that has been working fine for 2
In my experience using height for radius works, for example if you
have a 20 ft high ceiling then the coverage for one speaker would be
40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft
has never killed anyone, but this really depends on the power of the
speaker, I usually deal
Please DO NOT EVER CONTACT me or anyone off list if it's an answer to
something on list unless specifically asked to do so.
On Sat, Jun 26, 2010 at 10:43 PM, Thomas Perron thomas.per...@gmail.com wrote:
It kinda did not work.
exten = s,n,Set(CALLERID(name)=label${CALLERID(name)})
exten =
exten = s,n,Set(CALLERID(name)=label${CALLERID(name)})
put this before the dial command.
On Sat, Jun 26, 2010 at 10:09 PM, Thomas Perron thomas.per...@gmail.com wrote:
I want a call to connect via my DID to my dialplan.
Then, I want to attach a label to the incoming call
call arrives
starts
shows the
span as down.
On Sun, Jun 13, 2010 at 6:35 PM, C F shma...@gmail.com wrote:
Not sure what version you are running but I'm still running 1.2x in
1.2 you can't bring up PRI outside asterisk, since the PRI (I'm
assuming layer 2+) part loads with Asterisk.
On Sat, Jun 12, 2010 at 10
One more thing, read the comments here:
http://www.voip-info.org/wiki/index.php?page_id=573tk=2ff846f8169b7694aed5comments_page=1
Don't forget to have a beer ready :P
On Mon, Jun 14, 2010 at 7:18 PM, C F shma...@gmail.com wrote:
Your configs looks good, the only thing left is to figure out:
1
Not sure what version you are running but I'm still running 1.2x in
1.2 you can't bring up PRI outside asterisk, since the PRI (I'm
assuming layer 2+) part loads with Asterisk.
On Sat, Jun 12, 2010 at 10:51 AM, Voip Asterisk aster...@wideideas.com wrote:
Ya i'm not even to the asterisk part yet.
Interface type?
On Fri, May 28, 2010 at 1:47 AM, bhrugu mehta mehtabhr...@gmail.com wrote:
hi, all
Is ther any way to set up call-waiting feature in asterisk using dialplan or
any other ways. I want to use only
asterisk for that not any other gui.
I am using asterisk 1.4.28.
Regards,
I don't think you are actually hitting the time out. Comment out the
set timeout line I think the results will be the same. Which tells me
the timeout is not kicking in.
On 4/29/10, Brendan Sterne bren...@callvine.com wrote:
Greetings,
I'm trying to continue to do some processing after a
If you use zap then asterisk already does it. With sip the phones will
not tell asterisk about the hook flash. However you can play around
with dynamic features and assign a key that will mimic hook flash.
Injecting the beep sound might be hard though. Playing a different
ring to 2nd caller based
Viking electronics analog phones (I think E20) connected to an ATA.
Or cyberdata door boxes.
On Mon, Apr 12, 2010 at 1:19 PM, Mark Hulber asterisk.ad...@hulber.com wrote:
I want to use a voip speaker phone as a room monitor. Requirements:
A phone that I can set to auto answer in speaker mode.
Does a regular phone work on that port of the channel bank?
On Wed, Mar 17, 2010 at 5:00 PM, Robert Grignon rgrig...@fleetone.com wrote:
I have two Viking E10 Door phones and a Rhino FXS channel bank...
I have the channel set to immediate=yes and defined a custom context...
When I press the
Anyone know of a cordless 4 line DECT phone that does't have an
answering machine? Or one that costs less than $300.00 (even with
answering)?
Yes I tried Google but G too many results still sifting thru.
TIA
--
_
--
, C F shma...@gmail.com wrote:
Anyone know of a cordless 4 line DECT phone that does't have an
answering machine? Or one that costs less than $300.00 (even with
answering)?
Yes I tried Google but G too many results still sifting thru.
TIA
On Mon, Mar 1, 2010 at 5:16 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Mon, 1 Mar 2010, C F wrote:
Anyone know of a cordless 4 line DECT phone that does't have an
answering machine? Or one that costs less than $300.00 (even with
answering)?
4 line? Do you mean 4 handsets or 4
Wouldn't some online translator do a better job? Or just plain old
spell checking?
On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE cipr...@carsenie.ro wrote:
If anyoane have a firmware with sip support for a tainet venus 2804 please
give am feedback caz i kan-t find on internet
--
In 1.2 you can use rtp debug in the CLI
On Thu, Feb 25, 2010 at 8:27 PM, Alejandro Recarey
alexreca...@gmail.com wrote:
I'm trying to get my asterisk server to reinvite. I have two asterisk
servers with public IP's. My users (behind NAT) register on one server
(I'll call it server 1), and for
It should use the context of the device
On Wed, Feb 17, 2010 at 8:40 PM, Joseph syscon...@gmail.com wrote:
Is there any asterisk guru who can explain me how how asterisk knows which
context forward the call to?
--
Joseph
--
On Sun, Feb 14, 2010 at 2:30 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sat, Feb 13, 2010 at 09:25:01PM -0500, C F wrote:
Excellent and very informative article, Thanks Olle.
I ran thru lots of my dialplans now quickly to see if I have a catch
all exten anywhere. I couldn't find any
On Sun, Feb 14, 2010 at 3:26 AM, Olle E. Johansson o...@edvina.net wrote:
14 feb 2010 kl. 03.25 skrev C F:
Excellent and very informative article, Thanks Olle.
You're welcome.
I ran thru lots of my dialplans now quickly to see if I have a catch
all exten anywhere. I couldn't find any
While I like these solutions, they should never be substituting a good
secure dialplan.
On Sun, Feb 14, 2010 at 3:04 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Sun, 14 Feb 2010, Kyle Kienapfel wrote:
strip_ampersands(${EXTEN})?
(sip.conf)
[general]
allow-characters
Excellent and very informative article, Thanks Olle.
I ran thru lots of my dialplans now quickly to see if I have a catch
all exten anywhere. I couldn't find any that are accessible
unauthenticated, I always declare all fixed length extensions using
patterns the exception being international
On Fri, Feb 5, 2010 at 10:08 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 5 Feb 2010, Thomas Perron wrote:
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined
On Fri, Feb 5, 2010 at 9:54 PM, Thomas Perron thomas.per...@gmail.com wrote:
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined digit, the call gets bridged to me.
There is a
Sorry Thomas, but I have to agree with Karl on this one.
We have been here long enough to smell whats behind a posters motives.
There are the guys that make a living out of this one way or another,
usually selling a usefull service. Which is quite noticeable on the
type of posts those people make
.
While some posts of you showed that you havn't yet figured out how to
use asterisk but you do see the potential it has in creating a
business, the answers given back then should have pointed you in the
right direction if you were really truly an academic person.
On Sat, Feb 6, 2010 at 10:30 PM, C F
For a case like this I would go with overhead paging.
On Fri, Feb 5, 2010 at 4:50 PM, Mark Willis marksli...@markwillis.net wrote:
Has anyone done any large scale intercom deployments with Asterisk? I've
been asked about building a system to one-way page 500 phones
simultaneously from a
remember the settings but if I had a terminal in front of me, I am
sure I could get it work.
Thanks,
Steve T
On Wed, Jan 27, 2010 at 5:42 PM, C F shma...@gmail.com wrote:
We didn't fix it yet. For the moment the Definity is not connected
directly to Asterisk, we route all communications between
a mantis bug https://issues.asterisk.org/view.php?id=16713 and
hopefully we can get this issue resolved.
Alec
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Thursday, 20 August 2009 11:21 a.m
On Fri, Jan 22, 2010 at 2:11 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hey hey!
Anyone got any subjective (!) views on the merits of these two ranges
, using asterisk 1.4 ? I need to supply approx 30 handsets to a new
client, with the senior managers (6) having
ROFL.
Thanks, was waiting for this one.
On Fri, Jan 22, 2010 at 2:22 PM, Randy R randulo2...@gmail.com wrote:
http://twitpic.com/z8n36
On Fri, Jan 22, 2010 at 8:11 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hey hey!
Anyone got any subjective (!) views on the
On Fri, Jan 22, 2010 at 3:06 PM, Andrew Latham lath...@gmail.com wrote:
I have worked on many snom phones over the years I have never had
a snom phone go bad...
I guess you are not long enough in this business.
I have repaired stuck screens and overheated sticky bits but all in
all
Want them? I have them.
On Fri, Jan 22, 2010 at 3:16 PM, Tim Nelson tnel...@rockbochs.com wrote:
- Andrew Latham lath...@gmail.com wrote:
Having demo phones is priceless. Sometimes I show off the phones I
like with some phones I don't like to show the end users why it
matters... Bad
On Sun, Jan 24, 2010 at 1:23 PM, C F shma...@gmail.com wrote:
On Fri, Jan 22, 2010 at 2:11 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hey hey!
Anyone got any subjective (!) views on the merits of these two ranges
, using asterisk 1.4 ? I need to supply approx 30
Anyone on the list ever used it?
I'm trying to quote a system with 192 analog ports, one of the options
are the Xorcom 32 channel FXS USB Channel Banks.
Any input would be appreciated.
TIA
--
_
-- Bandwidth and Colocation
Any echo issues using FXS ports?
On Tue, Jan 12, 2010 at 6:53 PM, Carlos Chavez cur...@telecomabmex.com wrote:
On Tue, 2010-01-12 at 18:05 -0500, C F wrote:
Anyone on the list ever used it?
I'm trying to quote a system with 192 analog ports, one of the options
are the Xorcom 32 channel FXS
Anyone using the above mentioned SIP Gateway made by grandstream?
I would like to hear some feedback on real life experience using this gateway.
TIA
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On Tue, Jan 5, 2010 at 9:40 PM, Shane Brath sh...@brath.net wrote:
Folks,
I have a Merlin Legend R7 V10.0 with a 2 100D cards.
Sorry, I feel your pain.
I have 1 card in slot 4 going to CenturyTel, and the card in slot 10 going
to a flip cable to a TE110P card in a Asterisk 1.6.x box.
I
On Sun, Jan 3, 2010 at 9:14 PM, Nicholas Blasgen
nicho...@refractivedialer.com wrote:
I'm trying to move my Asterisk deployments under a Virtual IP address and
now remember why I dislike this. My primary Asterisk system is now behind a
firewall in private address space. My question is what
Before I start I am a Panasonic certified dealer AND I have installed
over 100 Asterisk systems that are in production.
That said for your application use Panasonic, DONT use Asterisk.
Use the Panasonic KX-TDA50G. Supports up to around 50 ports.
In addition to Analog and their proprietary Digital
On Mon, Dec 28, 2009 at 5:42 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Rick Huebner wrote:
My brother-in-law is finishing up his McMansion and I've done all of the
low voltage wiring and am starting the trimout. We are batting around
what to do for a phone system and I'm torn
On Mon, Dec 28, 2009 at 11:45 PM, Doug d...@natel.net wrote:
At 16:13 12/28/2009, Rick Huebner wrote:
My brother-in-law is finishing up his McMansion and I've done all of the
low voltage wiring and am starting the trimout. We are batting around
what to do for a phone system and I'm torn
Huge thanks for mentioning what type of channel you are using.
On Tue, Dec 22, 2009 at 5:11 AM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi C F,
I solved the problem!! It was under my nose...
If you are interested the solution is here:
http://www.misdn.org/index.php
You would have to create a dialplan for it.
If your provider expects *67 (which is the case here with I/CLEC POTS)
then you would create something like:
exten = _*67[2-9]XX,1,Dial(Zap/g1/${EXTEN})
In the case of PRI you would use:
exten = _*67[2-9]XX,1,SetCallerPres(prohib)
exten =
On Thu, Nov 19, 2009 at 10:31 PM, aster...@opensourcesolution.in wrote:
hello friends i want very simple thing in my dial plan.
1.When ever calls come at exten 2000 and if it is not answered with in 60
secs it should hangup.
Set absolute timeout to 60 seconds.
2.when ever call comes at
can you define are not working?
I just tried it on my cell phone and doesn't work either. Probably
because ATT didn't define them.
2009/11/5 Torintino T torinti...@hotmail.com:
I found and *65 are not working.
Please how can i re-enable them again.
Thanks
On Wed, Oct 28, 2009 at 10:57 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
C F thankyou very much.
when i make a call to Asterisk server recieves and works fine. But as to
make external calls we have to press nine so supposed a logic to dial 9
first then wait and then dail other
Any simple legacy integration will work. Search on voip-info.org
Here are some problems that I know exist with panasonic systems on
their SLT (analog) ports:
1. No CPC, Asterisk if connected using station ports on the TDA to FXO
on asterisk, will not detect hangups since the TDA will not send
Try this link:
http://tinyurl.com/yl7ra6w
On Wed, Oct 14, 2009 at 6:26 PM, Dan Journo
d...@keshercommunications.com wrote:
Already checked there, however it doesn’t really give a great deal of
information.
Many other links?
Thanks
Dan
From: asterisk-users-boun...@lists.digium.com
You mind elaborating what you mean by cloud?
On Tue, Oct 13, 2009 at 8:22 PM, Dan Journo
d...@keshercommunications.com wrote:
Hi,
I was wondering if anyone is successfully running Asterisk in a cloud
environment.
If you could state which cloud you are using, I’d appreciate it.
Many
are you using chan_local?
try disabling the hardware DTMF.
Sent using my wired Blueberry.
On 10/9/09, nik600 nik...@gmail.com wrote:
Dear all
i have a TE205P connected to an Asterisk 1.2.18.
Yes i know, the version is old but since now the system was stable and
i don't have the necessity
The local channel always generates 2 calls, one for the original call
and another for the proxy local chan.
2009/10/8 Anahi Ludueña a_ludu...@hotmail.com:
Hi people,
I have the following dialplan, but it doesn't have the behavior that I think
it should have.
[default]
exten = 2001,1,Answer
Couple of old posts:
http://lists.digium.com/pipermail/asterisk-users/2007-April/186195.html
http://lists.digium.com/pipermail/asterisk-users/2009-March/229479.html
http://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html
On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis
On Mon, Sep 28, 2009 at 11:51 PM, Alec Davis siva...@paradise.net.nz wrote:
I'm interested, and I expect others will be on how you might use it.
Our use on the mantis bug, is to allow 3 ISDN connected sites (no reliable
internet) each running asterisk, to dial other staff members in the other
On Sat, Sep 26, 2009 at 12:47 PM, Alex Balashov
abalas...@evaristesys.com wrote:
Don't put a SIP server behind destination NAT. Just don't.
Why not? Mind to explain?
ABBAS SHAKEEL wrote:
Sorry My Question was not very clear.
Asterisk System that is placed some where on local LAN (suppose
Sorry but AIPHONE is a terrible choice for this.
On Thu, Sep 24, 2009 at 8:53 AM, Chris Mason (Lists) li...@masonc.com wrote:
AIPHONE makes all that stuff, I would not try to reinvent that.
Vincent wrote:
Hello
I assume I'm not the first one to think about this: Is it possible to
connect
I have done something similar using the following:
1. An Adit 600 with FXS card.
2. A door box from Viking
http://vikingelectronics.com/products/view_product.php?pid=428
3. An inline dialer from viking:
http://vikingelectronics.com/products/view_product.php?pid=137
4. A relay activated using an
mistakes.
On Tue, Sep 22, 2009 at 11:31 AM, Barry L. Kline blkl...@attglobal.net wrote:
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C F wrote:
I have seen lots of companies offering this as a service and have used
phonetag.com in the past.
They work very nicely, however I have a customer
How are these identical?
On Tue, Sep 22, 2009 at 11:32 PM, Julian Yap julianok...@gmail.com wrote:
I have an issue where a particular dialplan works but another doesn't. I'm
not sure why. To me they look identical and it has me stumped.
This works:
[to-test]
exten = _X., 1,
I have seen lots of companies offering this as a service and have used
phonetag.com in the past.
They work very nicely, however I have a customer that is not
interested in paying $30-$40 a month but would rather buy the
software. I have googled and googled all I can come up with are
companies that
and how are those POTS lines connected to Asterisk?
In any event doing something like:
Set(CALLERID(num)=${CALLERID(num):0:10}) should do the trick.
On Tue, Sep 8, 2009 at 12:27 PM, Jeremy Taylor jer...@getwiredright.com wrote:
Hi,
I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones.
I need to add a second T1 to an asterisk system. However the first
card is in a PCI-e slot, and the only available slot is a PCI card.
Could that work?
TIA
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 -
Thanks all for your responses. It happens to be that there will be no
bridging whats so ever between the 2 cards.
Thanks again.
On Thu, Aug 20, 2009 at 4:01 PM, Dave
Fullertondfullertaster...@shorelinecontainer.com wrote:
C F wrote:
I need to add a second T1 to an asterisk system. However
Which PRI timer will fix this?
On Thu, Aug 20, 2009 at 3:58 PM, Dave
Fullertondfullertaster...@shorelinecontainer.com wrote:
C F wrote:
We have setup asterisk to handle our calls before between telco and an
Avaya definity. The PRI keeps locking up every so often.
In addition I keep getting
We have setup asterisk to handle our calls before between telco and an
Avaya definity. The PRI keeps locking up every so often.
In addition I keep getting this error when trying to call the avaya:
-- Channel 0/2, span 1 got hangup request, cause 102
-- Hungup 'Zap/2-1'
When that error
I have a provider that in order to set outbound CID they want me to
make sure that the From Header in the sip invite matches the caller ID
while the contact header matches the registration info.
For example.
My phone number with my provider is 2125551212 which is also my
username. I want caller ID
You have to pay LD rates.
On Fri, Jul 17, 2009 at 1:42 AM, Alex Balashovabalas...@evaristesys.com wrote:
C F wrote:
If you don't want to port it to the PRI for whatever reason you can
convert it to a RCFW (remote call forwarded number) which is around
$15.00 plus $8.00 for each additional
In the good old days telcos didn't care how many channels your forward
used up, they just did it. However nowadays they only allow one
channel at a time to be forwarded, if you need more you have to pay
for it.
Verizon here in NJ charges around $8.00 a month for each call path
(channel), and so do
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