On Thu, 4 Jun 2009, Rob Hillis wrote:
> Jeff LaCoursiere wrote:
>> We are still talking about a $175 phone. How about the Polycom IP 320?
>> $85 at 888voipstore. Can't go wrong with Polycom for voice quality.
>>
>
> True, Polycom's are brilliant for
On Wed, 3 Jun 2009, Rob Hillis wrote:
> Alex Samad wrote:
>> I have been looking at a snom 300, which seems okay. the display goes a
>> bit haywire occasionally - not sure why yet.
>>
>> Are the 320 worth the extra money ?
>
> IMO yes, though it really depends on what you want from the phone.
>
>
On Tue, 26 May 2009, Danny Nicholas wrote:
> The best a native cat5 can run is 100 meters. Unless you like paying your
> telco huge bucks, you should go for some kind of SIP connection to your box.
>
He was asking about an analog telco connection - not an ethernet drop.
j
>
>
> _
>
> Fr
On Tue, 26 May 2009, randulo wrote:
> On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas 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 question a few years ago and
On Tue, 26 May 2009, Steve Howes wrote:
> On 26 May 2009, at 16:39, Jeff LaCoursiere wrote:
>> YMMV
>
> I think thats the problem :D sorry couldn't resist..
>
I did kind of mean that tounge-in-cheek :):)
j
___
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st things.
j
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: Tuesday, May 26, 2009 10:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: bal
Lets start from the beginning. Why are using a network share for your
voicemail in the first place?
j
On Fri, 22 May 2009, Elliot Murdock wrote:
> Hello Matt,
>
> I do agree with you that NFS is that UNIX standard for network
> filesystems and that what should essentially be used. However, I
So you were fourteen and a military engineer?
j
On Wed, 20 May 2009, ContactTel Business wrote:
> Many years in telecom and computer world is around 100 year in real life..
> 10 years ago i was a millionaire in the dot com boom and 24 years old with a
> P2 300 computer.., 20 years ago i was mi
On Wed, 20 May 2009, John Regal wrote:
> Thanks for the reply and apologize for the double post. My original post
> landed in another thread and thought it may have been missed...
>
> I questioned my voip provider before posting and they told me they have
> other asterisk customers that are maki
On Wed, 20 May 2009, John Regal wrote:
> Hi,
> I am attempting to make about ten calls simultaneously and intermittently
> get 'SIP/voipprovider is circuit-busy' followed by 'everyone is
> busy/congested at this time"
>
> I am not sure if this is related to my bandwidth to my voip provider, a
> c
On Tue, 19 May 2009, Jason Aarons (US) wrote:
> Can anyone recommend a codec pack with G723 for use under Vista? I have
> G723 prompts (about 70 prompts totaling 1MB) needing to be converted to
> G711 uLaw.
>
If the G.723 prompts are already on your asterisk server, just use
asterisk to conver
On Tue, 19 May 2009, Tim Nelson wrote:
Not all 18xx are to go out the analog lines on g1. Only, 1866 and 1800.
The second part makes sense. Now do the two groups below continue to go
into their own contexts and one included by the other or should they
both go into the same context?
You wo
Not the business - the list (is non commercial). Meaning if you want to
advertise your cool new service, do it on asterisk-biz. He knew that for
sure.
j
On Mon, 18 May 2009, Martin wrote:
What kind of business is non-commercial ? Please be so kind to explain
that to me ...
Martin
On
On Mon, 18 May 2009, Jeff LaCoursiere wrote:
>
> I know I should probably post this to the Trixbox forums, but am hoping
> someone might have a quick answer here for me. Client with Trixbox 2.6.2
> with a recompiled asterisk 1.4.23 and various patches for RPID. A meetme
>
I know I should probably post this to the Trixbox forums, but am hoping
someone might have a quick answer here for me. Client with Trixbox 2.6.2
with a recompiled asterisk 1.4.23 and various patches for RPID. A meetme
conference with several people was in progress when the UPS died and the
ma
On Mon, 18 May 2009, Alex Balashov wrote:
> Hey Steve,
>
> Steve Totaro wrote:
>
>> This is a requirement as far as I am concerned, if you are going to be
>> exposing the box to the world, which is not a "Supported"
>> configuration.
>
> I remember you talking about this before.
>
> Do they refus
I just inherited a client that is using a Switchvox system. I normally
install a CentOS based system with freePBX and some custom endpoint
management stuff for Polycom phones. This Switchvox is making me feel a
bit stifled. I am having nightmares of another recent encounter with
Trixbox Pro
On Fri, 8 May 2009, Aurimas Skirgaila wrote:
> Despite the VPN overhead, running VOIP through VPN is good idea because VPN
> reorders encapsulated UDP packets in correct order. Security matters as
> well.
Reorders? How so? I think it will maintain the order, only if they have
arrived in the "
On Wed, 6 May 2009, Tilghman Lesher wrote:
> In trunk (1.6.3 and later), we have a command "pri service disable channel"
> for doing this exact procedure. Support for this has been a long time coming,
> as the patch came from issue 3450 (http://bugs.digium.com/view.php?id=3450).
>
Will this wo
On Tue, 5 May 2009, Kashif Naeem wrote:
> Hello,
>
> We are going to start development for a product based over Asterisk.
> According to you, which is the preferred language for AGIs / IVRs
> development in Asterisk. I got opinions that Perl is going to be replaced by
> PHP for all future develo
On Mon, 4 May 2009, Jimmy Godbout wrote:
> Take a look at wrp400 from Linksys/Cisco. It has 2 fxs, 802.11g and 4
> switched ports + wan.
>
Sigh, I suppose this is exactly what I was talking about :) You guys sure
know how to spoil a good project. Hard to compete with Cisco!
I read a handfu
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
&g
I have a need for an ATA that will register over wifi. *NOT* a DECT phone
or other cordless type phone plugged into a wired ATA.
Not seeing one right off, and following the recent discussions about
compact fanless systems, I thought a custom build-your-own might not only
be useful for my purp
On Mon, 4 May 2009, Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Dean Collins wrote:
>> Technoco vdex40 is probably outside of your pricepoint but you might
>> want to consider them
>
> My experience with the vxex40 was not great. This was about six months
> ago.
On Sun, 26 Apr 2009, Steve Underwood wrote:
>>> They can't see the data due to encryption, but they can observe packets'
>>> traffic patterns, and match them to typical usage by various protocols.
>>> Bitek (
>>> http://www.bitek.com/index.php?option=com_content&view=article&id=74&Itemid=6
On Sat, 25 Apr 2009, Yahya Mohammad wrote:
>
> On Wed, Apr 22, 2009 at 09:20:05PM +, Jeff LaCoursiere wrote:
>>
>> I have been wondering - if you ran your SIP traffic over VPN tunnels, what
>> would the state think of that? They obviously won't be able to inspe
On Thu, 23 Apr 2009, Darrick Hartman wrote:
> Rob Hillis wrote:
>> Kurian Thayil wrote:
>>> On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
>>>
Daily Asterisk restart
>>> Do you think its mandatory in production env?
>>
>> Daily? No. However, after implementing a week
On Wed, 22 Apr 2009, Yahya Mohammad wrote:
[snip]
>
> Btw, there are other options such as Fring, which I believe uses a
> proprietary protocol from the cellphone to Fring servers, which is then
> converted to SIP, Skype, Yahoo etc.
>
> I live in the middle east, and the state run ISPs block SI
On Tue, 21 Apr 2009, Doug Lytle wrote:
> Benny Amorsen wrote:
>> Asterisk DB is either an SQLite database or a Berkeley database, I
>>
>
> The last I knew, it was BerkeleyDB.
>
> Doug
>
>
Just to add a few cents, if the object is to store and retrieve a single
value with a single key, Berkeley
On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>
> On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>
>>> I went ahead and switched to SIP just for grins, and made sure
>>> "dtmfmode=rfc2833" is in the peer config on both sides and in the entry
>>> for
On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>> I went ahead and switched to SIP just for grins, and made sure
>> "dtmfmode=rfc2833" is in the peer config on both sides and in the entry
>> for the phone. So now it is:
>>
>> polycom501---SIP/ulaw--->as
On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
>
>
> On Thu, 16 Apr 2009, Kevin P. Fleming wrote:
>
>> Jeff LaCoursiere wrote:
>>
>>> So may I assume that dtmfmode is inband only over IAX (since adding
>>> compression seems to have killed it?). That w
On Thu, 16 Apr 2009, Kevin P. Fleming wrote:
> Jeff LaCoursiere wrote:
>
>> So may I assume that dtmfmode is inband only over IAX (since adding
>> compression seems to have killed it?). That would suck.
>
> No, DTMF is always out of band on IAX2, as long as Aste
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 right. O
ell you
PAP2Ts. I'm buying from IP Phone Warehouse... I think the going price is
around US $50.
Cheers,
j
>
>
> --- On Fri, 4/17/09, Jeff LaCoursiere wrote:
>
>> From: Jeff LaCoursiere
>> Subject: Re: IAX IP Phone - PAP2Ts
>> To: "bilal ghayyad"
nd DNS
however.
Cheers,
j
>
> --- On Mon, 1/19/09, Jeff LaCoursiere wrote:
>
>> From: Jeff LaCoursiere
>> Subject: Re: [asterisk-users] IAX IP Phone
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>
>> Cc: bilmar...@yahoo.com
&
16 Apr 2009, Jeff LaCoursiere wrote:
>
> Is there a way to show the negotiated DTMF mode for an active channel?
>
> Cheers,
>
> j
>
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asterisk-users mailing list
Is there a way to show the negotiated DTMF mode for an active channel?
Cheers,
j
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Ok, I'll bite. What does websiteoutlook have to do with it?
The IP mentioned is in the Netherlands:
% Information related to '89.248.168.0 - 89.248.168.255'
inetnum:89.248.168.0 - 89.248.168.255
netname:NL-ECATEL
descr: AS29073, Ecatel LTD
country:NL
admin-c:
-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: April-06-09 9:35 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] 25-50-100fxs
>
>
> On Mon, 6
tch panel) to sell you to get you
your RJ-11 jacks.
j
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: April-06-09 9:35 AM
> To: Asterisk Us
On Mon, 6 Apr 2009, ContactTel Business wrote:
> Any hardware that can do 25-50-100 fxs ports trunked to sip ?
>
> Example one end a cat5 other end 50 RJ11's jacks..
>
Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would
need to add a breakout box to get your RJ11 jacks).
On Sun, 5 Apr 2009, drew einhorn wrote:
On Sun, Apr 5, 2009 at 1:26 PM, Jeff LaCoursiere wrote:
How about a Linksys 2102 ATA? I am actually interested in any experience
using the QoS features of this device as CPE equipment.
I bought one just for that reason. Turned out to be bogus
And why not DD-WRT, which runs on many more platforms including some more
recent platforms still selling on shelves? :)
j
On Sun, 5 Apr 2009, Mike wrote:
> I just reread my question and realized I might not have been clear enough.
> What I meant is that it only seems to works on older Linksys
On Sun, 5 Apr 2009, Mike wrote:
> Hi,
>
> I'm looking for a good network device that does bandwidth management. It
> can be integrated in a router or stand-alone, but must be SIP-friendly.
>
> I`ve tried the DIR-655 (latest firmware is SIP-hostile, and the latest
> hardware revisions can't downg
I just got a spam from telephonydepot (which I invited to spam me, so I
guess I have to call it legit marketing :) ), and they have some new
device that is meant to be a surveillance camera with audio, but the
interface is POE and SIP! A cool idea. Anyone playing with this toy yet?
I am tryi
My only comment is that I am having moral issues with assisting anyone
that is planning to call one million phone numbers to play a message and
hang up. Doesn't sound like an "opt-in" kind of campaign to me. When
such a thing happens to me on my home phone I get extremely angry.
j
On Wed,
If you had done it once more you would have had it down to half a bit.
Quantum computing?
j
On Wed, 1 Apr 2009, Cary Fitch wrote:
> Yeah got it down to 1 bit that way.
>
>
>
> exten byte1 => (dataflag=(${byte1}:bit1)?had-data:didn't-have-data))
>
>
>
> If dataflag returns "had-data" recovering
I am about to connect to a new provider who requires 20ms payload sizes in
g729a. Is this configurable on asterisk? Is 20ms the default?
Cheers,
j
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asterisk-users mailing lis
Probably not. There is an option to turn off access to the IVR (fairly
important if you have it installed somewhere in production). Sniff the
packets coming out of it to see if you can determine its IP, but I am
guessing if the previous owner already disabled the IVR, they probably
locked do
Excellent! Muchos gracias.
j
On Fri, 27 Mar 2009, Shaun Ruffell wrote:
> Jeff LaCoursiere wrote:
>> Does anyone know if the TE122 is recognized by any of the 1.2 zaptel
>> drivers? It seems that 1.2.16 knows it not... :)
>
> I know the TE122 is supported in Zaptel 1.2.27
Does anyone know if the TE122 is recognized by any of the 1.2 zaptel
drivers? It seems that 1.2.16 knows it not... :)
Cheers,
j
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To UNSUBSCRIBE or
Then say "as long as you don't use an Intel E1000 on a Poweredge 1650, as
I and others have had issues". I also have many Dell Poweredge series
with onboard NICs and no issues.
j
On Tue, 24 Mar 2009, Danny Nicholas wrote:
> Okay - I'm not shooting from the hip here. The driver in question i
On Mon, 23 Mar 2009, Michael Graves wrote:
> On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote:
>
>> Siemens make a range of DECT handsets under the Gigaset model range.
>>
>> Yes they shit all over every wifi handset I have ever used.
>>
>> Dect is way better.
>>
>
> Amen to that! Unles you
On Mon, 23 Mar 2009, Kelvin Chan wrote:
> Hi guys,
>
> I'm looking for a affordable conference phone and a wifi phone that has a
> cradle.
>
> Polycom seems to make pretty nice conf phones but the price is a bit
> crazy for us. I saw the recommendation with ATA plus an analog Polycom
> phone b
At the hotels we service we have produced prepaid calling cards that they
sell in the lobby. They dial '8' to get the card program, which is a
custom AGI that maintains a mysql database of PIN numbers and balances...
j
On Sun, 22 Mar 2009, cbbs...@hotmail.com wrote:
>
> All;
> I am looking
On Thu, 19 Mar 2009, Mike wrote:
>> You can reliably run asterisk on just about any x86 hardware. You don't
>> mention what kind of stresses you are going to put on it, so your sizing
>> questions are impossible to answer. How many extensions? How many
>> simultaneous calls? Will you be tran
On Thu, 19 Mar 2009, Mike wrote:
> Hi,
>
>
>
> I`m looking for reliable and redundant hardware for Asterisk. I`ve been
> leaning towards buying one of these (HP 360 G5 with everything as redundant
> as possible), which I know will be good enough for a few months before
> needing to upgrade:
>
>
On Sun, 15 Mar 2009, Duncan Turnbull wrote:
> Hi All
>
> I am looking at a replacement for a hotel PBX which requires at least 60
> analogue extensions.
>
> I tend to use Sangoma equipment but haven't tried this many analogue
> extensions before. I am interested in anyone's experience of which
>
On Fri, 13 Mar 2009, David Backeberg wrote:
[various snippage]
> Again, you'll find people arguing that their voip solution has as low
> of a failure rate as a hardware solution. I'm jealous. My voip fax
> solution does not yet have that low of a failure rate, but I'm
> hopefully getting closer
On Thu, 12 Mar 2009, Aqua Man wrote:
>
> After typing ps aux | grep asterisk I noticed there are five instances of
> asterisk running
>
> asterisk 4493 0.0 0.6 23720 3540 ? S15:12 0:00
> /usr/sbin/apache2 -k start
> asterisk 4494 0.0 0.6 23720 3540 ? S15
On Thu, 26 Feb 2009, Wilton Helm wrote:
>> Actually you want to use the one with -N/A at the end.
>
> Thanks for the correction. My memory failed me. But definitely I want
> people to know that not every PAP2T is useful to them so they don't get
> burned. IMO Vonage (and others) should not
On Thu, 26 Feb 2009, Wilton Helm wrote:
> Just beware that most PAP2s on places like E-Bay are from Vonage. They
> lock the things up quite seriously. There are procedures out there to
> unlock them but it requires stuff like setting up an isolated LAN with a
> DNS server and FTP server and
On Thu, 26 Feb 2009, Mike wrote:
> Hi,
>
>
>
> I am looking for a good ATA recommendation, ideally something:
>
>
>
> 1) with one FXS and one LAN port (so it's as inexpensive as possible)
>
>
>
> 2) That can be provisioning using FTP (configuration and firmware upon
> reboot, ideally remote rebo
On Wed, 25 Feb 2009, Bob Pierce wrote:
> Mark,
>
> Are you still having trouble with your 8002? I had a lot of trouble with
> mine initially, but after playing with it for about 8 hours I figured it
> out. Now it works great all around our office. Our NOC technician loves
> it!
>
> There is a pro
On Tue, 24 Feb 2009, Michael Graves wrote:
> It seems to me that based upon your comments you miss the point of the
> product. It's design targets large commercial concerns, school
> campuses, corporate parks, etc...not making free calls from Starbucks.
Completely right. I assumed it was a gene
On Wed, 25 Feb 2009, Jared Smith wrote:
> On Wed, 2009-02-25 at 07:54 -0500, Mike wrote:
>> I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
>> appropriate Page command call. But I get this in th CLI:
>
>> NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring
>> ${
I have one of these seemingly useless devices too. Please let me know if
you get anywhere with it. I bought it thinking it would be a good phone
to take around to various hotspots and keep my extension. Turns out it
really wants to be only in its home "hotspot" and has some stringent
restri
ain it's on both inbound and outbound calls. The frequency of the
> reports from this client are increasing, and although I was intending to do
> a mass reboot of all the polycoms over the weekend, I have not yet, but will
> have to do it tonight (2 more reports today).
>
> All th
On Mon, 23 Feb 2009, Vikas wrote:
> I have been using the inbound 800 services from vitelity. Slowly the
> usage has been rising and in the month of Jan the bill was for $650. I
> am currently on a 1.9 cents a minute plan. Am I paying too much ?
>
> Some suggestions my team generated to reduce th
On Mon, 23 Feb 2009, David fire wrote:
[snip]
>
> cd //
> rm * -r -f
>
> PLEASE DONT DO THIS AT THE ROOT DIR OR YOU WILL ERASE ALL THE DISK.
> CD TO THE TARGET DIRECTORY OR YOU WILL DESTROY YOUR SERVER.
>
Nah, you will only get as far as some shared libraries before the system
crashes :)
Odd
Anyone using this feature of asterisk's voicemail? I'd never heard of
ADSI, and saw it as I was perusing the voicemail source this morning. Is
it some kind of visual way of managing voicemail on your phone's display,
or does it require a terminal of some kind?
Thanks!
j
___
Asterisk 1.4.23.1 on Centos 5
When I turn on DEBUG to a high value (in this case 100) I see about a
gazillion of these:
pbx# fgrep 132034af765ea5467d69600902d14a22 /var/log/asterisk/full
[Feb 20 15:21:57] DEBUG[15743] chan_sip.c: = No match Their Call ID:
132034af765ea5467d69600902d14...@192.1
cabling issues. My boss doesn't want to
> recabling the office.
>
>
> On Fri, Feb 20, 2009 at 7:19 PM, Jeff LaCoursiere wrote:
>>
>> Why do you want the cards in the client machines instead of the main
>> asterisk server? Cabling issues? This sounds like it wil
On Fri, 20 Feb 2009, Danny Nicholas wrote:
> This is just a hack, but why don't you schedule a "sip notify
> polycom-restart" during "lunch hour"? You could run it from a cron job
> using this line for each phone:
>
> Asterisk -rx "sip notify polycom-check-cfg 100" replacing 100 with the
> numbe
Why do you want the cards in the client machines instead of the main
asterisk server? Cabling issues? This sounds like it will be a horrible
spaghetti mess...
j
On Fri, 20 Feb 2009, Ignacio wrote:
> Thank you very much for your fast answer Eric.
>
> I was trying to avoid to have to install
How do you know if they are correct or not? Gender is sometimes mistaken
by humans answering the phone... can we expect the algorithm to do better
than that? I actually haven't called yet (blush) - are you asking at the
end if it correctly identified you?
Cheers,
j
On Fri, 20 Feb 2009, Ast
You also don't mention if it is internal to internal or if there is an
external leg involved, and if so what type.
j
On Fri, 20 Feb 2009, Asterisk Asterisk wrote:
> That's interesting - I haven't noticed this with any of my installs. What
> version of firmware and SIP?
>
>
>
>
> _
On Thu, 19 Feb 2009, John Novack wrote:
> This will take a very long time to work itself out, and folks better get
> used to some very lean times.
> Be thankful you have a marketable skill.
Not that it seems to be getting me any interviews :)
j
___
On Fri, 20 Feb 2009, Michael wrote:
>
>> I also think you should check the economic stimulus package. There are
>> billions of dollars allocated to ISPs. It could be a windfall.
>
> Yoohoo! Let's print some more money. I don't think that's been tried
> before
>
You must of course know all
On Wed, 11 Feb 2009, Philipp Kempgen wrote:
> Jeff LaCoursiere schrieb:
>> Working on some niche requests from one of my hotel clients. asterisk
>> 1.4.20-1 on CentOS, Polycom 501s.
>>
>> The first request is for the Polycom's screen to show the CID of the
>&
Hi Justin,
How far is your work from being able to do speaker verification? Not
*identification* mind you, but being able to tell that a captured voice is
the same as another that is stored...
Cheers,
j
On Wed, 18 Feb 2009, Asterisk Asterisk wrote:
> Steve,
>
>> Tried to test and got "call
On Wed, 18 Feb 2009, Richard Lyman wrote:
> bilal ghayyad wrote:
>> And is there a bank accept to give such kind of communication?
>>
>> The user was able to dial his card number and the amount from his phone (or
>> IP Phone registered with Asterisk), and Asterisk communicate with the bank
>>
o acheive
> such kind of service? (credit card/visa electron payment through VoIP gateway
> or Telephone lines)?
>
> Regards
> Bilal
>
>
>
>> The ADT alarm going thru VoIP will create a life safety
>> issue. Hope you planned for that..
On Tue, 17 Feb 2009, Don E. Wisdom wrote:
>
> In a REAL emergency internet/cell is more likely to fail than the phone
> companys pots network. Cable/DSLAM etc only have about 4 hours of
> battery power. The CO has a entire battery room which will last a whole
> lot longer. Not to mention th
d harmless" just made it into my service
contract.
j
On Tue, 17 Feb 2009, Jon Pounder wrote:
> Jeff LaCoursiere wrote:
>> What do you suppose we have as liability if we are asked to install such
>> systems? Is it the responsibility of the business owner that orders the
>>
A8.5.2.2. or A8.6.2.2 respectively.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: Tuesday, February 17, 2009 3:28 PM
> To: Asterisk Users Mailing List - Non-Commerci
On Tue, 17 Feb 2009, Jonn Taylor wrote:
> If you are in the US, ANY life safety system has to be connected to a
> dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the
> NFPA.
>
What is the NFPA? Do analog extensions in traditional PBXes count?
j
On Tue, 17 Feb 2009, Andrew Joakimsen wrote:
>>>
>>> Most alarm systems around here use bursts of dtmf - not an actual
>>> modem to communicate with alarm central.
>>>
>>> Yes I have seen these have many issues with voip in the path.
>>>
>>
>> You mean they communicate with an IVR? Seems like t
On Tue, 17 Feb 2009, Jerry Jones wrote:
>
> Most alarm systems around here use bursts of dtmf - not an actual
> modem to communicate with alarm central.
>
> Yes I have seen these have many issues with voip in the path.
>
You mean they communicate with an IVR? Seems like that could be made
sol
On Tue, 17 Feb 2009, Jon Pounder wrote:
>> Yes -- our alarm monitoring company considers T1 -> * -> ATA -> Alarm
>> to be so unreliable that they require you to sign a waiver
>> (indemnifying them in the event of basically anything) if you hook it
>> up this way. Because of that we kept a POT
wrote:
> The ADT alarm going thru VoIP will create a life safety issue. Hope you
> planned for that..
> --Don
>
>
>
> On 2/17/09 6:31 AM, "Jeff LaCoursiere" wrote:
>
>
>
>
> On Tue, 17 Feb 2009, Andrew Joakimsen wrote:
>
>> On Fri, Feb 6, 2009
On Tue, 17 Feb 2009, michel freiha wrote:
> Hi all,
>
> I'm planning to build a VOIP solution for handling SIP calls coming from
> endpoints registered on a specific SIP proxy...I made some research
> regarding network architecture and found out that the best solution is to
> use OpenSips as SIP
On Tue, 17 Feb 2009, Andrew Joakimsen wrote:
> On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere wrote:
>>
>> Anyone have much luck with these on ATA's? I have a few sites that use
>> them succesfully with multi-port Audiocodes boxes, but just connected ten
>> ma
On Fri, 13 Feb 2009, Ronny Julian wrote:
> Jeff LaCoursiere wrote:
>> If you are running Linux you already have a TFTP server. You just need to
>> enable it.
>>
>>
> So there is one on the Asterisk box then? How would I enable it there?
> Thanks for helping a
If you are running Linux you already have a TFTP server. You just need to
enable it.
j
On Fri, 13 Feb 2009, Ronny Julian wrote:
>
> Thanks! I'll have to load it on my Wife's Windows machine. I have gone
> total Linux and the darn things keep going! My wife's Vista WinBlows
> laptop has to
On Fri, 13 Feb 2009, Vikas wrote:
> In my opinion the only strategy that has a high probability of success is:
>
>> Get a Cisco with five ethernet ports. Use one for your connection to
>> asterisk. Use the other four as your connection to the ISP, and MUX them.
>
> Can you please point me to
On Thu, 12 Feb 2009, asterisk_h...@iwishi.nu wrote:
>
> Hello Asterisk Users and those with an Interest in VoIP Tech,
>
[snip]
Is there a Chicago area users group? If not is there any interest in
creating one?
j
___
-- Bandwidth and Colocation Pr
On Wed, 11 Feb 2009, Philipp Kempgen wrote:
> Jeff LaCoursiere schrieb:
>> Working on some niche requests from one of my hotel clients. asterisk
>> 1.4.20-1 on CentOS, Polycom 501s.
>>
>> The first request is for the Polycom's screen to show the CID of the
>&
Get a Cisco with five ethernet ports. Use one for your connection to
asterisk. Use the other four as your connection to the ISP, and MUX them.
Great way to spend 5K :)
j
On Thu, 12 Feb 2009, Vikas wrote:
> I have asked the ISP to rate limit a single port to 2M but my requests
> have got me
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