On 15-06-15 08:48 PM, Matt Darnell wrote:
In the past we have used Adtran Atlas 550's to break out FXS ports for
devices like modems. The great thing about the 550 is that internally
it is all TDM so there is absolutely zero latency.
We are able to use ATA's for faxes and analog phones but
On 12/04/2013 10:22 AM, Gregory Malsack wrote:
Its beyond disgusting. If it was not for legacy garbage nothing from m$
would be left in my datacenter.
Saying you are an expert Linux user is just a joke when you don't
understand the poor architectural choices that come with windows and why
it
On 07/18/2013 09:56 AM,
jacob.e.mi...@l-3com.com wrote:
I am attempting to setup my server to use
Lua for the dialplan (extentions.lua), but I am unable to get
the asterisk configure script to find the
On 07/14/2013 03:12 PM, bilal ghayyad wrote:
check ebay there are lots of wide power injectors with 4 8 or 16 ports.
Hello;
We have a cisco switches but they are not PoE and we need only to have
PoE device so the cables come for it first to provide the power and
then goes to the switch (to
On 04/09/2013 01:44 PM, Daniel - Asterisk wrote:
sqlite is not really a multiuser dbms, so really its hard to build tools
that do what you want without causing problems in the way it operates.
You'd be better off running postgres or mysql (yuck I said it), and
using one of the many tools
On 02/19/2013 11:20 AM, Christopher Harrington wrote:
I was always under the impression you needed to either use a cellphone
type device to send them using your account, or send on to one of the
aggregators who have apis for this.
For low volume stuff, you can simply send an email
On 01/11/2013 12:20 PM, A J Stiles wrote:
I try to write comparisons as != where possible and then there is no
confusion and less mistakes possible.
Most compilers will warn on the example below now.
On Friday 11 January 2013, penguin wrote:
quick question that leaves alittle confusion
On 01/09/2013 01:49 PM, Steve Edwards wrote:
I was about to reply 'no' but thought to check my spam logs so now I
reply 'yes.'
I got a few of them actually.
--
_
-- Bandwidth and Colocation Provided by
On 01/09/2013 03:52 PM, adriano wrote:
Might just mean operators working for the company that connect with
voip to the system and then take calls.
(the old way of doing this was centrex in a hunt group and people taking
calls at home)
I think that the mobile operator as any other company
On 01/07/2013 03:41 PM, Doug Lytle wrote:
The blowing fuses could be related to spikes etc., from a poor
connection to the source, or a problem with the source hardware.
If the amps are good, you could just drive them from a cheap phone
with a regular headset jack
They aren't, seem to be
On 01/02/2013 12:20 PM, Steve Totaro wrote:
good one - me too !
On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote:
On 1/2/2013 11:30 AM, Richard Kenner wrote:
If things were properly trimmed, the email would be short enough that it
really doesn't matter that much if
On 01/02/2013 03:22 PM, Patrick Lists wrote:
On 01/02/2013 06:20 PM, Steve Totaro wrote:
I became a list member way before any such rule and never had to click
through and agree to these update ToS.
I am grandfathered in.
Just looked it up. I see my first post back in April 2003, yours in
On 01/02/2013 03:35 PM, Jim Lucas wrote:
On 01/02/2013 12:16 PM, Don Kelly wrote:
I don't think Outlook does what I'd like, so I'm not limiting my
options. I
can use different email to keep track of the Asterisk lists.
Thunderbird (by default) bottom posts. And it does the nice indenting
On 12/30/2012 03:54 PM, Benny Amorsen wrote:
Boy what an elitist attitude.
I have been on this list far longer than most people - long before
digium even existed and if you don't value what I have to say - well
just don't read it.
If you or your mail reader can't slice and dice a mailing
On 12/28/2012 08:13 PM, Steve Edwards wrote:
Please stop saying don't top post, some of us prefer it that way.
Please don't top-post. If you don't know what that means, please
consult Google.
On Fri, 28 Dec 2012, bilal ghayyad wrote:
I have one more question:
What was u meaning by call
On 11/28/2012 11:52 PM, Steve Totaro wrote:
You're not serious right ?
That is just the center of the country since no better location is
available.
On Wed, Nov 28, 2012 at 7:45 PM, J Gao j...@veecall.com wrote:
This morning someone tried to make sip call through my Asterisk. My server
just
On 11/15/2012 10:27 AM, Eric Wieling wrote:
What I have found most difficult in any failover situation is having
everything decide at the same time something has failed.
(this applies to anything not just asterisk)
For example how does the polycom react if it can make the sip
connection, but
On 11/15/2012 10:31 AM, Danny Nicholas wrote:
ran into this before on routers, you can put something like that or vrrp
or carp in front of a pair of systems to fail to the right one BUT there
isn't only one interface on something like a pbx, it has a lan interface
and a wan interface, you
On 10/31/2012 02:49 PM, Jeff LaCoursiere wrote:
On 10/31/2012 01:44 PM, Russ Meyerriecks wrote:
On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote:
Anyone manage to make one of these work *on* an asterisk server?
Have been researching most of the morning and have only found
On 10/31/2012 02:38 PM, Jeff LaCoursiere wrote:
why not just get a usb headset and use with one of the sip client apps ?
if you're going to the trouble of having a phone to plug in the fxs why
rely on the pc at all ?
use one of the spa type routers and plug the pc into it and the phone
or if
On 10/25/2012 04:21 PM, Justin Killen wrote:
just talking in general terms here I have found this sort of hardware is
not the most reliable, and the more physical devices you spread it
across the more fault tolerant you are of a single fault taking down a
big chunk of your users.
I wouldn't
On 10/25/2012 05:09 PM, Carlos Alvarez wrote:
On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen
jkil...@allamericanasphalt.com
mailto:jkil...@allamericanasphalt.com wrote:
Cost and ease of deployment, yes. At this specifc location we are
currently using Centrex lines (ATT hosted) and
, Oct 25, 2012 at 4:29 PM, jon pounder j...@inline.net wrote:
On 10/25/2012 04:21 PM, Justin Killen wrote:
just talking in general terms here I have found this sort of hardware is not
the most reliable, and the more physical devices you spread it across the
more fault tolerant you are of a single
We are looking to find someone that is familiar with Fujitsu and Mitel
PBX's. Email ru...@inline.net off list.
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
Well, that means opening up VPN connections from everywhere. Thats why
I suggested turning off the server completely.
hmmm - I thought that was the point of a vpn
--
_
-- Bandwidth and Colocation Provided by
On 11/30/2011 09:01 AM, Tom Browning wrote:
I agree - its a bad comparison of 2 different things meant for different
purposes.
iptables is enforcement, fail2ban is detection.
if you have time to sit and make up iptables rules by hand during every
hack attempt
1) you have too much time on
On 11/25/2011 06:39 PM, Michelle Dupuis wrote:
There is a script on www.generationd.com designed for Asterisk. It will
convert the Wav49 to mp3, add call info into MP3 tags, add a company logo, etc.
and then email the message.
It's a one line change to add to asterisk - very handy. (We use
On 10/09/2011 09:52 AM, Silverthorne Wystead wrote:
wget each of the screens
I think there are only 6
Hello Folks;
I may be posting this in the wrong list, but here goes.
I have a Grandstream GXP2000 and I would like to use tftp or some
other utility to grab the configuration from it.
On 09/25/2011 04:41 PM, Alex Balashov wrote:
Sometimes people get such swelled heads they need a slap back to reality
- I completely agree with him the changes were idiotic.
Obviously the comments touched a nerve with you or you would not have
replied.
On 09/25/2011 02:23 PM, Bruce B
On 09/25/2011 08:47 PM, Bruce B wrote:
This is becoming just like the bacula mailing list where anyone that
knows anything is beaten into submission for daring to question the
great and powerful oz.
You are very childish besides being very useless.
Also, note that there are others that
On 08/26/2011 02:02 PM, linux guy wrote:
get any cheap android device and load linphone.
or grandstream works for a wired device.
gxp2000 has enough line buttons you can easily route calls for multiple
people to a phone and tell who the call is for
I'm looking for 4 to 6 good, inexpensive
On 08/26/2011 02:18 PM, linux guy wrote:
In our house, we need wireless. I have a Grandstream already.
I am looking for something with a form factor more conventional than a
cellphone. Maybe that is silly ? I see various unlocked large screen
Android devices for ~$150.
not sure what you
On 08/26/2011 02:26 PM, Jeff LaCoursiere wrote:
On Fri, 2011-08-26 at 12:10 -0600, linux guy wrote:
I was thinking of using a PAP2T-NA for the ATA to handle the fax. It
appears to have a large number of fax specific settings. Can anyone
comment on using this device with a fax ?
If you are
On 08/26/2011 03:17 PM, linux guy wrote:
I like the idea of running multiple ATAs with a single base or handset
on each line.
Something like the Panasonic KX-TG4111B which sells for about $40 for
a handset and base. PAP2s sell for about $50 or $25 per line. Total
cost of $65 per handset.
On 08/22/2011 04:11 PM, Linuxguy123 wrote:
I have a home and business system that just ties all the lines together
(combo of zaptel, and sip incoming at several locations), inbound
routing based on which line it came from. Was using a t1 card and
channel bank for extensions but migrated away
Personally, I have been shot at on top the Iraqi Government building
in the IZ from the Red Zone. I was setting up and troubleshooting the
Motorola Canopy WiFi system. Just a few 7.62x39 rounds, nothing I
would call heavy fire.
It was because you were setting up the canopy stuff, not
On 05/29/2011 09:37 AM, Michael R. Wally wrote:
So how long till its an adaptive telemarketing blocker based on the
query velocity of the numbers ?
The system uses real Telco CNAM Dips. Any generic names you get are
from the subscriber's carrier itself. We can only provide what we
On 05/24/2011 11:35 AM, A J Stiles wrote:
Someone asked about the quality of it, he was quoting the hardware specs
of a similar device.
I doubt magicjack publishes that kind of detail about theirs.
so where is the problem ? Its irrelevant he represents that device
commercially.
On
On 05/24/2011 11:57 AM, A J Stiles wrote:
On Tuesday 24 May 2011, jon pounder wrote:
On 05/24/2011 11:35 AM, A J Stiles wrote:
Someone asked about the quality of it, he was quoting the hardware specs
of a similar device.
No they didn't. The original message to which the spammer
On 05/24/2011 02:45 PM, Warren Selby wrote:
On Tue, May 24, 2011 at 12:33 PM, Doug Lytle supp...@drdos.info
mailto:supp...@drdos.info wrote:
Steve Edwards wrote:
My archives don't go back that far
Mine do.
No match on Jack, magic or itntelecom.com http://itntelecom.com
On 04/01/2011 11:00 AM, Dean Collins wrote:
Anyone on the list using and Android tablet with a voip service as
their primary phone device? Either with wired or Bluetooth headset?
What are you using hardware/software.
What are your thoughts?
I use linphone.
Config :
- incoming ivr etc to
On 03/03/2011 03:53 PM, Danny Nicholas wrote:
Not having an in-depth knowledge of how EU numbering works, I would
still suggest that you could get pretty far with the numberingplans
AGI if you made a database that blocked out the number once it came up
as a cell. In the U.S. cell phones
On 01/26/2011 08:52 AM, Gilles wrote:
If you like open source what are you doing running windows ?
Getting anything to work properly there which does network
communications is a huge PITA since every user has their own firewall
and different settings etc etc etc.
Hello
I'd
On 01/20/2011 12:01 PM, Andrew Thomas wrote:
why not just subscribe with an account that doesn't do that like gmail
or yahoo ?
Hi,
Is the any kind of 'tag' that I can include at the end of my message to
make the list processing software ignore and dispose of my disclaimer?
In other words -
Surely there is some mail client smart enough to be able to flip around
the levels of indenting so most recent is top or bottom.
If not quit bitching and make one - I will continue top posting since I
don't seem to be alone in preferring it.
On 01/16/2011 10:28 PM, Mark Murawski wrote:
We
On 12/04/2010 03:01 AM, Tammy A Wisdom wrote:
You all do know that this is something you could be sued for since this is
'life safety equipment' ? I've heard from multiple sources that if it isn't
against the nfpa that it will be very soon
Ask yourself if you want to be subject to a lawsuit
On 11/22/2010 06:44 PM, Kevin Keane wrote:
Use IPTables to lock down your machine to only accept incoming
connections from your local network and from the particular IPs that
you are expecting connections from (such as your SIP trunk, maybe).
That is of course assuming that these calls are
On 11/19/2010 10:10 AM, Michael Graves wrote:
On Fri, 19 Nov 2010 10:43:28 + (GMT), Gordon Henderson wrote:
Interestingly for commercial units, I've had the opposite experience -
I've found that my (business) customers just will not pay for something
tiny that's capable of supporting
On 11/18/2010 10:02 AM, Chris Gentle wrote:
On Tue, Nov 16, 2010 at 8:28 AM, Gilles codecompl...@free.fr
mailto:codecompl...@free.fr wrote:
Hello
For users who 1) don't have a QoS-capable ADSL router and 2) would
like to run Asterisk with a couple of SIP trunks, I was wondering
On 11/15/2010 02:49 PM, Mark Scholten wrote:
Anyone have a soft sip endpoint which can take touchtones over sip and
run scripts ?
that is a more general purpose integration solution to asterisk itself.
I realize there are scripts for dialplans which can do this already but
often the door is
I'm still on old-fashion copper-wire and have yet to experience the joy of
SIP Trunk-ing and the type of issues discussed in this thread. My thought
to share here is that outgoing calls should be easy for thoroughly
authenticated users and impossible for others...
Probably more
On 11/01/2010 01:44 PM, Nyamul Hassan wrote:
I think the only real solution here is to make people take more
responsibility for their actions
- find and punish the actual abusers
- make users liable for damages caused by infected PC's - defaults from
an isp should be everything locked down
unexploitable systems.
On Nov 1, 2010, at 11:54 AM, jon pounder j...@inline.net
mailto:j...@inline.net wrote:
On 11/01/2010 01:44 PM, Nyamul Hassan wrote:
I think the only real solution here is to make people take more
responsibility for their actions
- find and punish the actual abusers
- make
I already have a monitor (tied into nagios, which pages me if my fraud
thresholds are exceeded), but I feel that is probably beyond the
abilities of most of the people experiencing call fraud. The people
who know what they are doing with Unix and Asterisk are generally not
the victims
On 10/31/2010 11:39 AM, Mark Deneen wrote:
On Sun, Oct 31, 2010 at 11:26 AM, Joel Maslakjmas...@antelope.net wrote:
If these are mobile users, I hope they never use any public networks
(hotels, starbucks) where other subscribers can do things like ARP attacks
to do MITM (and steal your
On 10/31/2010 12:58 PM, Joel Maslak wrote:
On Oct 31, 2010, at 9:40 AM, jon pounderj...@inline.net wrote:
what are you using that is tied to nagios ?
I'll package it up next week and make it available.
Basically, I use nrpe to call a shell script that looks at the last five
On 10/30/2010 04:07 PM, Stuart Sheldon wrote:
any registry of abusers like for spam ?
any list of complete ip ranges for countries where abuse is rampant to
block ?
I am getting sick of the one offs and ready to start blocking big chunks
of address space.
-BEGIN PGP SIGNED
On 10/30/2010 09:24 PM, Sebastian wrote:
On 10/29/2010 04:40 AM, jon pounder wrote:
On 10/28/2010 11:18 PM, GBR Icasiano, Ryan A. wrote:
Here is what I do today and it works fine:
- asterisk/trixbox
- Dext/android phone
- Bell Canada cell provider
- call comes in, to an extension
On 10/30/2010 11:25 PM, Warren Selby wrote:
To me it seems the real question is What is going on today?. I normally get
eight to ten asterisk-related fail2ban alerts a day between a few client
sites - today I've received at least 10 times that many attacks on just one
site. These are all
On 10/28/2010 11:18 PM, GBR Icasiano, Ryan A. wrote:
Here is what I do today and it works fine:
- asterisk/trixbox
- Dext/android phone
- Bell Canada cell provider
- call comes in, to an extension with voicemail
- rings a bunch of sip devices (real phones, and the android via
linphone if it
On 10/02/2010 02:56 PM, bruce bruce wrote:
Hi Everyone
I think PAP2T supports DynDNS and other Dynamic DNS providers. I have
a box that needs to be secured at all times. Currently it's not
connected to the internet. If it were connected, I would have iptables
block any and all traffic
://mybox.dyndns.org
localnet=192.168.0.0/255.255.255.0 http://192.168.0.0/255.255.255.0
every time the address changes you have to have some script to make the
change in your firewall.
???
Thansk again,
On Sat, Oct 2, 2010 at 2:59 PM, jon pounder j...@inline.net
mailto:j...@inline.net wrote
On 09/16/2010 12:01 PM, Chris Owen wrote:
well that just means you need a trunked satellite pbx where all the
phones are, and that would take load off the main connection.
half those people have got to just be talking to each other and don't
need to use the gateway at all.
On Sep 16, 2010,
On 09/15/2010 12:42 PM, Leif Madsen wrote:
On 10-09-15 05:25 AM, Jonas Kellens wrote:
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx
On 09/12/2010 02:34 PM, Kyle Kienapfel wrote:
Really it depends on what the capabilies of dsl were assuming you are
just using both dsl and t1 as internet connections.
a dsl that has close to 1mb/sec out and 10mb/sec or so in, is going to
be pretty comparable to a t1 actually so not really
SIP wrote:
what can you do ? simple discard spam don't bounce it.
On 7/28/10 9:45 PM, Sam wrote:
Just a note, the asterisk mailing list server continually gets
blacklisted over at
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
mail to spamtraps. Perhaps
Mike Diehl wrote:
Hi all,
I've cross-posted this to the -users and -biz groups. Hope that's OK.
I have a customer who REALLY needs to be able to send/receive faxes reliably.
I could probably get hylafax configured, but I'm not sure how reliable it is.
If it is considered reliable,
Douglas Pasqua wrote:
Hi People,
I work in a company that are using asterisk as pbx.
I need a way to identify what client my employees are calling. For
example:
- For each call that an employee of my company make to a customer,
must identify the client name in the CDR table.
- Is there
Mark Willis wrote:
This could potentially create a very weird audio situation where the
delay between adjacent phones is audible so instead of acting like
loudspeakers in parallel on a conventional system, it just sounds like a
bunch of people talking at once and is not understandable.
Has
Richard Kenner wrote:
Is there a way to make a virtual extension busy programmatically?
I want to be able to turn lights on and off on a Polycom phone from a script.
That's what the Custom device type is for.
please elaborate I would like to know too
--
Kingsley Tart wrote:
try several fax machines and see if you get the same results
Hi,
I'm trying to receive faxes using hylafax / iaxmodem but I just can't
get it to work. We're using Sangoma E1 cards and have calls coming in
over PSTN. I've tried turning hardware echo cancellation off but
Kyle Kienapfel wrote:
Going along the internet between us and canada doesn't add much
distance, but bouncing back and forth between east and west coast
does.
I'm not so sure that is the case, what I do know is both Rogers and Shaw
can never seem to fix complaint issues with voip unless
Randy R wrote:
I might try a live cd once or twice, or use it to boot a dead computer
or one that is not mine, BUT for anything with any sort of time
investment in settings to try anything you lose it all with a live cd so
why bother since if you can't try it all in one session, you have to
Travis Elsberry wrote:
Hello all,
Do you know if it IS possible to use multiple lines/extensions on SIP
with a Cisco 7960 or other phone models? My boss wanted to have 1
physical phone but have it register to a couple of different
extensions, then use different ringtones to identify
Dean Collins wrote:
Earlier in the thread someone made a comment about using gsm since
everyone had gsm handsets already.
Can you explain in detail please ? (what hardware specifically, and how
does this actually work ?) My ignorant assumption is something like the
end user has a cell phone
Benny Amorsen wrote:
Jared Smith jsm...@digium.com writes:
Not that I would ever consider taking an exam like that, but I have been
using/configuring asterisk since nearly the beginning of this mailing
list, and I have never touched dahdi or polycom. Someone should still be
able to pass
Tony Mountifield wrote:
I posted a message to this list about 50 minutes ago. I received a
posting acknowledgement pretty quickly, and it showed up in the mailman
list archives, but I still have not received a copy.
Looking at some of the other recent messages I have received, they have
also
D. Dante Lorenso wrote:
part of this is making a statement to get publicity, if twitter really
didn't like what you were doing they'd simply cut off your app accessing
their servers. But obviously that is not what its about.
Dean Collins wrote:
I received this email 30 minutes ago
Carlos Ruiz Diaz wrote:
Hello list,
Why PC modems were not used as FXO devices? Why chan_modem was
deprecated? it seemed a nicer option instead of buying expensive gateways.
the digium single fxo cards and clones for about $10 ARE modems.
you can get a sip gateway fxo + fxs in one box for
hongkong and china - tons on ebay and
auctions in all sorts of currencies with nearly free shipping, so yes it
is cheap, does not matter where you live, you just need to look.
Thanks.
On Sun, Aug 2, 2009 at 3:07 PM, jon pounder j...@inline.net
mailto:j...@inline.net wrote:
Carlos Ruiz
Jerry Geis wrote:
oh you mean a telemarketing pest server ?
I was wondering if (2) quad T1 cards
will work nicely in 1 server with a quad core AMD 3.0 gig cpu?
Basically used to dial out and deliver messages. play wav files for the
message.
Any thoughts.
Jerry
David Backeberg wrote:
On Thu, Jul 9, 2009 at 6:34 PM, Jerry Geisge...@pagestation.com wrote:
I was wondering if (2) quad T1 cards
will work nicely in 1 server with a quad core AMD 3.0 gig cpu?
Yes. Buy a server that has the corresponding ports to accommodate the
cards. A modern
Diogo Saad wrote:
Using an ATA, Do I still need a softphone or it´s embedded in the
hardware?
plain old walmart phone plugs in the ata (with or without callerid,
adsi, cordless, etc)
On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com
Miguel Molina wrote:
randulo escribió:
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote:
I run my analog telco over cat5, but that's in-house and definitely not
3km. That sounds really far for current loop stuff.
I was doing that too. I asked this same
Wilton Helm wrote:
one thing I missed mentioning about fxs devices - the linksys/sipura
ones actually allow you to set line characteristics on the slic inside
it. you can vary from the 600ohm default, and tweak gains a bit. Some
mix of a capacitive line or different resistance may help. never
, May 26, 2009 at 1:19 PM, Jon Pounder j...@inline.net
mailto:j...@inline.net wrote:
Diogo Saad wrote:
Using an ATA, Do I still need a softphone or it´s embedded in the
hardware?
plain old walmart phone plugs in the ata (with or without callerid,
adsi, cordless, etc
Wilton Helm wrote:
You are exactly right. Cat 5 had no advantage over cheaper wire for
voice, and the length limitations are meaningless. Consider that Cat 5
is typically use with signals that extent to 30 MHz or beyond. A voice
grade analog circuit must go to 4 KHz (1/10,000 as much). At
asterisk-us...@rogg.is wrote:
couple last words on this - if that is the application, then ringing the
remote terminal may not even be necessary, you really only care about
the hookswitch and audio which is a different thing entirely from ringing.
You may be able to boost the battery voltage
John Novack wrote:
If this is an emergency phone situation then I would question the wisdom
of even considering using Asterisk.
Conventional telephony solutions exist that will easily cover the loop
length and provide the reliability that should be required by risk
management in such a
Jonathan Moore wrote:
On Wed, May 6, 2009 at 10:53 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Not sure how you would do that, as the X100 card is an FXO card, won't
provide either battery or dial tone to the cordless.
What you will want for that is an FXS card or ATA.
The X100
Jeff LaCoursiere wrote:
why not just put something like a wet11(wireless bridge) and pap2t(2x
fxs) in the same box ?
dev time = 0
cost ~100
those are just two of the many products that would work together to do
what you want.
I have a need for an ATA that will register over wifi. *NOT* a
Jeff LaCoursiere wrote:
On Mon, 4 May 2009, Jon Pounder wrote:
Jeff LaCoursiere wrote:
why not just put something like a wet11(wireless bridge) and pap2t(2x
fxs) in the same box ?
dev time = 0
cost ~100
those are just two of the many products that would work together to do
what you
Eric Fort wrote:
Anyone know where I could find a good beginning for using asterisk and
the text based game adventure together such that I could play from
the nearest phone?
google on collossal cave.
honestly its the absolute worst unreadable mess of code ever conceived
by man or beast.
Jeff LaCoursiere wrote:
On Fri, 17 Apr 2009, bilal ghayyad wrote:
on the subject - I have one and it runs VERY hot - is this normal ? I am
almost afraid to leave it on its so hot.
Thanks a lot, but I need IAX IP PHone.
About the PAP2Ts, in which price u r getting it? Any link that u
Jeff LaCoursiere wrote:
On Fri, 17 Apr 2009, Jon Pounder wrote:
Jeff LaCoursiere wrote:
On Fri, 17 Apr 2009, bilal ghayyad wrote:
on the subject - I have one and it runs VERY hot - is this normal ? I am
almost afraid to leave it on its so hot.
That doesn't sound
Erick,
how about posting your home phone number here so we can all call you and
play a 20second audio clip - I am sure you would see nothing wrong with
that would you ?
ContactTel Business wrote:
Your right, i don't think we would help someone asking on advice to send 1
million emails for
Roger Marquis wrote:
Steve Totaro wrote:
I understand you are a developer and you want IAX2 to be great.
That is your job, but the fact is that it is not and has caused
audio and security problems for YEARS in EVERY release. It
should bug you and everyone at Digium that waves the IAX2
David Ruggles wrote:
I've done some googling and searched voip-info but I'm not able to find a
good answer about how to provision the GXP 2000.
Based on questions I've asked before it seems like a lot of people are using
the grandstream phones so I figure provisioning can't be that hard. Is
Vieri wrote:
I see much the same except I think if you investigate further, the light
will be green whether the phone ever registered or not.
--- On Tue, 3/24/09, Ken Williams k...@intermountainelectronics.com wrote:
Our work around is to lower the
registration expiration on the phones.
Frank Bulk wrote:
In a SOHO environment I would agree with you, but not if your coverage area
needs to be tens of thousands of square feet. Deploying a complete overlay
wireless infrastructure doesn't make sense and is another infrastructure to
manage and maintain.
did you think about
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