Let's get back to the highly valid point that Robert made. He is
absolutely right. Good parents teach their children to do the right
thing regardless of what the majority of children are doing these days.
Consider 2 approaches to dealing with a compromised PC. I think the
majority of the
Adam Robins wrote:
I recently switched from BroadVoice to VoicePulse Connect on my Asterisk
box. Too many Meetme quality complaints (whether real or perceived).
I had to make a choice to use IAX2 or SIP with VoicePulse. I first
tried to go with SIP because I already had it working and all of our
,Congestion
Paul
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,Congestion
Paul
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I have setup the menu system, it works fine, but I can't get it to forward
the call to another outside number. The sites you gave me are on setting up
the IVR. Any thoughts?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett
Sent: Wednesday,
Any and all help is appreciated at this point. Thanks for the tip. This is
the only thing I have not been able to get working and ironically it is the
most important.
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mitchel
Constantin
Sent: Wednesday
How do I transfer or forward a call that is in the IVR and
connect it to a static external phone number? The call would come in on a POTS
line into an x100p and could go out using 3-way calling or another POTS line
connected to another x100p. Please help!!!
Paul
success
that I've had is using the DTMF command and manually dialing the number. My
phone rings, but asterisk hangs up before I can answer it. I could set a
timeout on it, but it would disconnect the call after that time even if I
was still on the phone. Please help!
Paul
if this is
possible and if so, how it's accomplished?
Paul
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An Asterisk box without window$ is like a chocolate cake without mustard.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian C.
Fertig
Sent: Friday, April 01, 2005 11:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
An Asterisk box without window$ is like a chocolate cake without mustard.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed
Sent: Friday, April 01, 2005 11:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
OK, so how does asterisk know how to bridge the two calls together and also,
what is the syntax to dial another number is extensions.conf?
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, April 01, 2005 14:35
How do I dialout using an extensions.conf and connect to an outside number?
For example, I would like for a person in the IVR to be able to press a
number and it dial out using another FXO card and another POTS line and then
bridge the two calls together.
Paul
for this
from your extensions.conf???
Cheers,
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Friday, April 01, 2005 17:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Preserving CallerID
Matt wrote:
Hi,
I'm currently routing my asterisk server out over broadvoice.. it
seems I can do multiple outgoing and incoming calls does anyone
know if broadvoice actually allows this or not?
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On Sat, 9 Apr 2005, Stuart Ford wrote:
Dear all ...
I'm experiencing terrible trouble with crackling and noise on an
analogue line connected to an X100P (compatible) card. I've checked the
line with a normal analogue phone and it works fine, clear as a bell,
but any outgoing or incoming calls
Rich Adamson wrote:
Sure you can, in most cases. Just check the fine print in their
service agreements, or whatever else they publish. If its not
their, call them as a prospective customer. If they don't answer,
then why bother to do business with them as that's going to be
about the same level of
to be able to pick up the handset, press 9, get a dialtone and
then dial the 10 digit number without having to press any softkeys. Does
anyone have any ideas??
Paul
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is
interfering with it. I thought maybe it was a wireless phone or router, so I
disconnected all those and put my cell phone in the other room. Still no
change. Anyone have any ideas?
Paul
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is
interfering with it. I thought maybe it was a wireless phone or router, so I
disconnected all those and put my cell phone in the other room. Still no
change. Anyone have any ideas, this is really getting to be a problem.
Paul
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Paul
From: Damian Funnell [mailto:[EMAIL PROTECTED]
Sent: Monday, April 11, 2005 12:49
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Line Noise HELP!
Hi Paul, there was a thread
for
the static...it never came. I wish the problem was less sporadic. Thanks
again for your post.
Cheers,
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damian Funnell
Sent: Tuesday, April 12, 2005 14:02
To: Asterisk Users Mailing List - Non-Commercial
Linn Boyd wrote:
I have looked for a quality IAX provider for 800 DID's we currently
have two, one is ok and the other is just not of quality, but last
night we got an email after a complaint of quality earlier in the day
and this is what it said. Remember I never did request a network
change,
Hi, I worked with Asterisk 0.7 without problems until I tryed to
loadH323. I downloaded the last version and after some try I compile
it.I followed the description in /asterisk/channels/h323/Readmeand
the compilation of this part was good. But the new compilationof Asterisk
was impossible
Now I try to download Asterisk 1.0.3 from ftp.asterisk.org/pub/asterisk
and it work again, perhaps the version in cvs has a problem with my
configuration...
only for information...
ciao ciao Paul
Paul wrote:
Hi, I worked with Asterisk 0.7 without problems until I
tryed to load
.
For those reasons I am desperately hoping that wifi sip phone prices
will rapidly fall.
Paul
James H. Thompson wrote:
Uniden and Vtech both just announced cordless
phones with SIP ATAs built into the base station.
You get better range and battery life compared to
a WiFi phone.
Jim
There is a sane and safe way to do this when you are not sure if the
patches are already in:
1 - create a new scratch directory
2 - copy the patch and chan_sip.c to that directory
3 - patch chan_sip.c broadvoicesip2.txt
If you have the right(unpatched) version of the c source you will only
get
claims that the patch was incorporated into the * cvs tree
as of 12/12/2004
--dalon
On Tue, 18 Jan 2005 13:15:01 -0800, David Shaw [EMAIL PROTECTED] wrote:
I ran a test patch like Paul said.
[EMAIL PROTECTED] test]# patch chan_sip.c broadvoicesip2.txt
patching file chan_sip.c
Hunk #1 FAILED at 231
:
Thanks Paul, This is over my head for now. I will sign up for broadvoice
next week and see if it works. I had signed up for Broadvoice last week
had problems so I drop them. They said I had connected but I couldn't
receive or place calls with them.
Thanks, David
It seems highly likely
Sometimes I have problems and changing to another of their servers makes
it start working again. There probably is a way to make * deal with this
properly. I am using the broadvoice account for test purposes at this
time so I just edit sip.conf and restart * when this happens. What I
have
for sip registration is sip.broadvoice.com I
have several for outbound proxy proxy.chi.broadvoice.com and etc...
Do you have any other for sip?
Best regards,
Helder
- Original Message -
From: Paul [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
A quick patch test yields a mixture of hunks failing and succeeding. I
guess my approach will be to build it and try it with broadvoice first.
I certainly don't want to spend any time analyzing code that is not needed.
I already have a 1.0.2 debian package set that works with broadvoice. I
to have a
cross-trained pbx management team. Or maybe you can get a good price on
this and bundle it with systems you sell. The screen shots on the
website didn't tell me enough to make me salivate over it.
--
Paul
Henry Devito wrote:
-Original Message-
From: [EMAIL PROTECTED
Start out with this:
BYOD-Lite
A special plan for our BYOD users only: BYOD-Lite for $5.95 a month.
You'll get your pick of numbers, all of our basic and advanced features
http://www.broadvoice.com/features.html, including 100 minutes of
outbound calls in the United States or Canada, AND the
This might be of interest:
I have a local DID and added a tollfree number. I was experimenting with
codecs so I changed the config to only allow g729. The tollfree number
stopped working but the local DID still worked. I allowed ulaw and it
all worked again. When I have time I am going to test
Quoting Gary Carr [EMAIL PROTECTED]:
You might want to tell that to these guys:
http://www.voipsupply.com/product_info.php?products_id=317
regards,
Paul
No, the PAP2's are. The PAP2-NA is for any provider.
Gary
- Original Message -
From: Steve Underwood [EMAIL PROTECTED
Keith O'Brien wrote:
Does anyone know where I can find the Asterisk Python interpreter?
The one referenced here:
http://www.sineapps.com/news.php?rssid=173
It seems like the site
http://vox.groovy.net/moin/PyAsterisk
is down
I just found something here:
Robert Webb wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Goodyear
Sent: Thursday, February 17, 2005 8:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anyone having trouble with
With unlimited calling plans you need to read the terms of service.
Sharing the account within a household or business usually fits in with
that. Reselling services in any way is usually prohibited.
Some providers with unlimited plans will allow you to set the outbound
caller ID to any number
Brian Capouch wrote:
Race Vanderdecken wrote:
Good, then let me move on to the insults and ranting.
1. Why are you running on Slackware? Are you trying to prove a
point or just enjoy being frustrated?
Open Source is like Broad Spectrum Pesticide, it works but
your results may vary and you
Andrew Kohlsmith wrote:
On March 5, 2005 08:14 am, Androtech wrote:
I bought one Trust 56k V92 PCI Internal Modem MD-1100 which has the 1057
Motorola Chip, and I installed it on my linux box.
When I try to load the module wcfxo, I cannot load it (zaptel is already
loaded):
Not to
Mike Dent wrote:
And if nobody's going to educate the newbies, then how will they ever learn?
Do you believe in letting your children do whatever they want, too? There
are 'defacto' rules for any system. No, I don't have my shiny ListCoptm
There is a difference when you are that childs
Andrew Kohlsmith wrote:
On March 5, 2005 02:46 pm, Paul wrote:
Maybe one of the free web-based forum packages will eventually offer an
elitist or impatient mode. Before you can post, you do the required
reading and pass online exams. The idea is to weed out people who think
README is just
Andrew Kohlsmith wrote:
I can't tell if you're on my side of this debate or not, because you sure
sound like it. :-)
I'm on my side and always on the side of the serious and earnest end
user. I live in a sparsely populated rural area. Most of the locals
can't afford to pay for computer
No, I want it to work 50% of the time and pay half your current pricing.
Or maybe we can make this really easy for you to understand. Make it
work 0% of the time and we pay you nothing.
I think that people expect it to work about 99.99% of the time if they
are going to use it for production
Steven Critchfield wrote:
On Mon, 2005-03-07 at 16:06 -0500, Eric wrote:
Hi Vinko,
MySQL blobs will store binary data, so you should be OK there. I'd
focus on whether or not storing the data in a variable is a good idea.
Typically, with any programming language, it's good practice to
keep
I too would like this.
Thanks so much
~paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terry
Bohaning
Sent: Thursday, January 15, 2004 9:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] vmail cgi script
Hi all,
I'm in the process of building
meetme.conf
; Configuration file for MeetMe simple conference rooms
; for Asterisk of course.
;
[rooms]
;
; Usage is conf = confno
;
conf = 1000
conf = 1001
Is there something more I need to do here?
Paul
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How do you enable music on hold?
From the little bit I found on the * site, it looks like all you do is
uncomment one of the lines in musiconhold.conf
What else is to be done?
~paul
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After installing mpg123 * will no longer start
up. I get the following error.
ERROR[16384]: File asterisk.c, Line 1349 (main): Unable to
connect to remote asterisk
If I remove mpg123, * will run as usual. Any ideas?
~paul
Has anyone had any success using the ztdummy module and
doing meetme/conferencing with out zaptel hardware installed?
Paul
) and asterisk, in both cases I receve errors
during make or make linux26 (I saw the notes on
http://www.voip-info.org/wiki+Asterisk+Zaptel+Installation).
somebody can help me?
thank you very much
Paul
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from Honk Kong to another
VOIP gateway? In essense Asterisk would do no termination, only route
the IP packets to a SINGLE IP address.
Hope that made sense.
Paul Seniuk
Freestyle Networks
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: No such file or directory
Tiffiop.h is just not present on any partition. I am running RH ES 3.0
Paul.
-Original Message-
From: admin [mailto:[EMAIL PROTECTED]
Sent: September 20, 2004 8:55 AM
To: asterisk-users
Subject: Re: [Asterisk-Users] spandsp / compilation errors
Graham Turner
load_modules:
Loading module app_rxfax.so failed!
Ouch ... error while writing audio data: : Broken pipe
Any ideas what could be the cause?
Paul Seniuk
-Original Message-
From: admin [mailto:[EMAIL PROTECTED]
Sent: September 20, 2004 10:38 AM
To: asterisk-users
Subject: Re: [Asterisk
libspandsp.so is indeed under /usr/local/lib
How can I verify my lib path (feel stupid for asking) ?
Paul Seniuk
-Original Message-
From: steveu [mailto:[EMAIL PROTECTED]
Sent: September 20, 2004 10:54 AM
To: asterisk-users
Subject: Re: [Asterisk-Users] spandsp / compilation
' extensions in my
various
Contexts.
Hope that makes sense,
Paul Seniuk
Paul Seniuk.vcf
Description: Binary data
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What happens if I want it to work over the same DiD though?
Does Answer() take care of this?
How do I jump to the fax extension if it detects a faxtone?
Paul Seniuk
-Original Message-
From: mstorck [mailto:[EMAIL PROTECTED]
Sent: September 20, 2004 3:51 PM
To: asterisk-users
if it detects a faxtone?
Paul Seniuk
-Original Message-
From: mstorck [mailto:[EMAIL PROTECTED]
Sent: September 20, 2004 3:51 PM
To: asterisk-users
Subject: Re: [Asterisk-Users] Question about the 'fax' extension
If you use extension dedicated to fax, then you don't need to use
Joe Greco wrote:
quote who=C F
Why is their DNS failing?
Looks like ns1 is down. Probably their master DNS server.
ns2 is up, but looks like their zone expired, since it could not refresh
from ns1, so it is no longer reporting authoritative for nufone.net.
They should look into
The thread is about music on hold. Things such as playing local radio
stations in a waiting room are not related. I don't think there is
anything illegal about using normal over the air radio and TV for such
purposes as long as it stays in the local market area.
Stephen Bosch wrote:
John Novack
I have the same debian and asterisk version combo running in more than
one location. Some are T1 and some are in data centers. There have been
times when I got such messages and some simple ping/traceroute testing
showed obvious problems at my end or the provider end. Problems at the
provider end
MOSBAH ABDELKADER wrote:
Hello,
Have i to install OpenVPN in each Asterisk server or it is enough to
install it in one side only?.
Thanks.
You have to install it on both sides of the tunnel. You will have to
read some docs to get it configured and working. You can learn something
new or
Mike wrote:
In the interest of making things cleaner, I'd like to know if I can
just reload one single conf file. Let's say I have two files,
extensions.conf which includes small_file.conf.
I only want small_file.conf reloaded, not the main file. Is this at
all possible?
Mike
Saludos a todos. El caballo que me vendiste ya no sirve. Mi chevy es en
el shop.
Let's make this Spanish Day for the list. Tomorrow will be Latin Day
and the day after will be Ancient Egyptian Day.
Just kidding. For those in countries where today is a holiday - enjoy
it. For the others - take an
Does it switch back to wifi from gsm tower? If so, I would hope they
count total gsm seconds for the call to determine how many minutes get
deducted from the wireless plan. Otherwise, somebody could get clipped a
full minute for every time he leaned out the window to yell at the kids.
[EMAIL
Alexander Lopez wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul
Sent: Friday, September 21, 2007 10:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
Mike Diehl wrote:
I just had a user complain about a call getting dropped and another one
failing to go through.
I'm trying to interpret the log entries for each call and would like to
confirm my understanding.
The first entry is from an outbound call to a 1-866 number. The [EMAIL
Does anyone have * talking to a Syndeo switch?
I am new to * and trying to get this working.
It appears to have a problem authenticating, but I do not
see an unauthorized message.
Then shows the * as a registered end point with an ip of
0.0.0.0
@1073435488.955240 las-cms-a :366
Jean-Michel Hiver wrote:
Hi List,
I was wondering if anybody had tried running Asterisk inside
virtualization software such as Xen. Are there known problems doing it?
Cheers,
Jean-Michel.
I have been running several asterisk xen servers for a few months now.
Problems would depend on what
Scott Geist wrote:
How do you retreive the caller id on incoming analog lines and display
the id on the sip phones on the network?
Assuming the analog lines have caller ID feature:
Do you have an incoming SIP account to test with? Reason I ask is that
my IP phones and ATA do that by default.
The way to first test the ATA is with a phone or caller ID display that
supports caller ID with call waiting. Some devices work that way by
default and some might require you to set the option. That test should
have nothing to do with zapata.conf at all. I assume the reason you
mention
Wes Baehr wrote:
I have been trying to get a DID set up with Nufone for the past 3 weeks.
They replied to my original inquiries before I signed up for account,
but now they have not responded to any of my e-mails asking to get a
DID set up.
Maybe I am hitting their spam filters, or do
Maybe some people think that a PBX should come with a few games just
like so many cell phones these days :)
Technical Support wrote:
I think that some people try to make their asterisk box a do-everything
super server. Can you image a traditional PBX with direct access via the
internet, serving
Ryan Amos wrote:
This is turning into a sysadmin theory flamewar, but I think the main
point is that Fedora probably isn't the best thing to run on production
machines for QA reasons. This is because Fedora is more or less the QA
testbed for RHEL. CentOS is, for all intents and purposes (except a
The -biz list is more appropriate for this.
Tele Cost Price Reducer wrote:
Hi Sam Tam,
i would be interested in these ATA that you can offer.
please provide me with more details about this option.
thank you very much,
Mickey Lazar
On 2/9/06, *Sam Tam* [EMAIL PROTECTED]
Assembly instruction sets usually have more documentation.
Anyway, the OP has a hotmail address so he already has a web interface
to use for the list.
Douglas Garstang wrote:
Yes, programming the dialplan is akin to programming assembler.
-Original Message-
From: Anthony Rodgers
Rich Adamson wrote:
app_milliwatt is a nice tool for a quick check of the
line quality.
Anyway, hearing to that tone for more than a minute is
painful.
Does anyone know the opposite application, i.e. an
application, that hears and listens for a 1000 Hz
tone and displays the quality in any unit?
I have seen some very expensive switches fail. Nice thing about lower
cost devices is that you can afford to have spares. If you stick to a
standard way of labeling and connecting wires you can use good open
source monitoring software to detect switch failure. If you allow people
to randomly
Andrew Kohlsmith wrote:
On Friday 24 February 2006 07:56, Paul wrote:
Maybe the first approach should be to setup a test extension for
recording the tone. The idea is to get best resolution possible in real
time. Then process it as much as needed to get the info you want. Such
an approach
Zach A wrote:
Hi everybody,
This question is confusing me for some time. From selling point of view
to a customer, calling asterisk a PBX doesn't look right. According to
the definitions of PBX or PABX, Asterisk is not just PBX but much more
than that. My question is, how should I introduce
Martin Joseph wrote:
On Feb 26, 2006, at 12:57 AM, Alexander Burke wrote:
Hello, list!
After Googling and checking out the voip-info wiki, I haven't had
much luck in locating a decent web-based voicemail system for
Asterisk to check your VM while you're away from the office without
using
Dumpolid Exeplish wrote:
i am taking overr the administration of an existing production * PBX
but i cant seem to find out which version of * this is. When i use the
'show version' coomandat the cli, i get this:
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
In some situations you could execute agi by just adding it to an
extension on the other server that gets the transferred call. The
associated information could be passed by various means. That decision
would be based on criteria like the frequency and volume of these
transfers. A simple prototype
Stephen Bosch wrote:
Diego Iastrubni wrote:
On Tuesday 24 April 2007 16:24, Stephen Bosch wrote:
Well, I can't speak for anybody else, but I haven't had a problem with
reproducing a source install.
How about time?
2 minutes download+install, vs 10-20 minutes compilation.
Then,
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
abandoning SugarCRM, and opting to develop their own Visual FoxPro
database/CRM?
Please don't dump on me now, this is not my idea, I am just asking for
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
Has abandoning SugarCRM, and opting to develop their own Visual FoxPro
in changing.
On 4/30/07, *Paul* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Wrote: 4/30/2007
8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists
Richard Lyman wrote:
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system,
yet Has abandoning
Stephen Bosch wrote:
Paul wrote:
Third - I have enough exposure to Visual FoxPro to quickly rule it out
as a choice for anything new. The fact that somebody is proposing to
use it might give you the idea that they don't know what they are
talking about at all. BTW - my exposure to it did
Richard Lyman wrote:
Paul wrote:
Richard Lyman wrote:
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever
Eric ManxPower Wieling wrote:
Jon Pounder wrote:
that's what dry copper is supposed to be, just a cross connect
between 2
pairs out of the CO. ie not even battery, line test equipment, or
anything
else hanging off it at the CO. any restriction should be purely a
function
of the
Stephen Bosch wrote:
Per Jessen wrote:
Jon Pounder wrote:
Quoting Stephen Bosch [EMAIL PROTECTED]:
C F wrote:
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of
In that case your post about vmware has nothing to do with the problem
discussed here.
Adam Robins wrote:
Thanks, but we do not use any zap hardware in these systems. It is straight
SIP.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François
Ed Nuñez wrote:
Is anyone else having trouble going into voip-info.org today?
I didn't have trouble going there a few hours ago but they were
delivering blank pages.
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Erik Anderson wrote:
On 6/12/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
I don't know of any other GE phones.
However...
Why in the world would you ever need GigE sip phones?
Maybe for people who
I'm going to top post in this situation.
Kevin - Commands that operate on the channel variables won't help if we
are using a call file. We will have a new channel.
This syntax works with asterisk 1.2.x for me:
Application: AGI
Data: say_it.php|call_status_message
I have done other things where
Looks to me like the law only targets intentionally deceptive spoofing.
Dovid B wrote:
Anyone know if this is only to bother some one ? I have a client
that has a consulting business. The clients call in and his asterisk
server call's his cell when he is out of the office. It passes along
For interactive GUI-based editing, I have used audacity on linux
workstations.
I use command line sox for things such as format conversions.
Andrew Latham wrote:
sox
On 7/10/07, Gary Chen [EMAIL PROTECTED] wrote:
I recorded some sound files using Asterisk record() app as ulaw file. I need
Dovid B wrote:
Hi List,
Does anyone know where I can get support for the digium forums ? my
user ID and pass just stoped working as of yesterday. The forums say
to go to asterisk.org for any password issues. I am able to log in
there with out any issues. For some reason when I try to log in
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