Hi guys,
I am trying to compile zaptel, using debian 4r5. However what I get in
zaptel 1.2.27 after make is below :
You do not appear to have the sources for the 2.6.18-6-486 kernel
installed (under ).
make: *** [modules] Error 1
tried to change the source with zaptel-1.4.12.1
reza adinata wrote:
Hi guys,
I am trying to compile zaptel, using debian 4r5. However what I get in
zaptel 1.2.27 after make is below :
You do not appear to have the sources for the 2.6.18-6-486 kernel
installed (under ).
make: *** [modules] Error 1
tried to change the source with
asterisk:/usr/src/zaptel-1.2.27# uname -r
2.6.18-6-486
doesn't that mean that I have already got the precise version in my
box? (uname - r-kernel-release print the kernel release) ? why do I
have to install the same kernel?
thank you
On 2/17/09, Dave Cotton dcot...@linuxautrement.com wrote:
is any program , to manage freemin on sim cards ,for gsm gateways that
connected to the asterisk, for termination?
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On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote:
asterisk:/usr/src/zaptel-1.2.27# uname -r
2.6.18-6-486
doesn't that mean that I have already got the precise version in my
box? (uname - r-kernel-release print the kernel release) ? why do I
have to install the same kernel?
Hi,
Yes, it is indeed working. I am currently using a debian4r5, and i can
install using aptitude
The problem is that I am trying to install an asterisk mp3player using
mpg123 that is capable of playing from .pls. And in some literatures I
have read, it is mentioned that I should have the newest
Tzafrir Cohen wrote:
On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote:
asterisk:/usr/src/zaptel-1.2.27# uname -r
2.6.18-6-486
doesn't that mean that I have already got the precise version in my
box? (uname - r-kernel-release print the kernel release) ? why do I
have to install
On Tue, Feb 17, 2009 at 09:54:21AM +, bails wrote:
Tzafrir Cohen wrote:
On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote:
asterisk:/usr/src/zaptel-1.2.27# uname -r
2.6.18-6-486
doesn't that mean that I have already got the precise version in my
box? (uname -
On Tue, Feb 17, 2009 at 10:30:31AM +, Gordon Henderson wrote:
On Tue, 17 Feb 2009, reza adinata wrote:
asterisk:/usr/src/zaptel-1.2.27# uname -r
2.6.18-6-486
Just a minor issue here - there was an issue with kernels 2.6.18 whereby a
user could get root access by running a simple
On Tue, Feb 17, 2009 at 05:19:42PM +0700, reza adinata wrote:
i am sorry, but I am not using English as my main language.. A bit
confused with several explanations above :(
what i get is that :
asterisk:/home/tsp# aptitude install zaptel-source
aptitude not installed? Well, just use
i am sorry, but I am not using English as my main language.. A bit
confused with several explanations above :(
what i get is that :
asterisk:/home/tsp# aptitude install zaptel-source
bash: aptitude: command not found
asterisk:/home/tsp# m-a prepare
bash: m-a: command not found
On Tue, Feb 17, 2009 at 04:48:37PM +0700, reza adinata wrote:
On 2/17/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote:
asterisk:/usr/src/zaptel-1.2.27# uname -r
2.6.18-6-486
doesn't that mean that I have already got the
Hi,
There are 3 new settings (setinterfacevar, setqueueentryvar,
setqueuevar) and membermacro settings in 1.6 queues.conf. What is
the potential use of these settings? The variables set are useful, but
there is no indication of the purpose they could be used? Any one with
some light on
On Tue, 17 Feb 2009, reza adinata wrote:
asterisk:/usr/src/zaptel-1.2.27# uname -r
2.6.18-6-486
Just a minor issue here - there was an issue with kernels 2.6.18 whereby a
user could get root access by running a simple program. I'm not sure if
Debian patched it though, but it might be
Hi,
Has anyone met something like this ?
dialor*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
7541/7541 (Unspecified)D 0UNKNOWN
7540/7540 (Unspecified)D 0UNKNOWN
7534/7534
Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message.
Regards,
Marc
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: mardi 17 février 2009 14:06
To: Asterisk Users Mailing List - Non-Commercial
On Tue, 17 Feb 2009, Andrew Joakimsen wrote:
On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere j...@jeff.net 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
machines to Linksys 2102s and
2009/2/17 Marc STORCK msto...@voipgate.com
Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION
message.
Yes.
I think that simply, in this case, the endpoint (SIP phone) is just broken :
it wouldn't reply to anything ...
I'm not 100% sure now, but wouldn't be surprised ...
Did you use the same screen name / name for the 2 SIP extensions you setup
on the one phone? If so, some phones will confuse asterisk based on the SIP
header (in particular AASTRA phones). If this is an Aastra phone, this is
probably the cause...
From: asterisk-users-boun...@lists.digium.com
Ok isn't this replacing a western hack with a bridge hack? The init
0 and init 6 probably aren't going to work anyway since (1) asterisk has
to be running as root and (2) the path in * is limited if even existent, so
the init command would work unless you had a copy or symlink in the asterisk
Hi there,
We're having some complaints of choppy audio from our SIP customers.
Asterisk is showing no errors, but I'm getting a lot of these in my syslog:
Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654-8232)
The first number varies, but the last number is always 8232.
I've
This indicates that your NIC card is not handling the throughput
effectively. Is * the only application on your server? How many users are
on * when this occurs?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
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 proxy for registration and local calls between
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
No, asterisk on conventional hardware can handle at most a few hundred
calls.
I would strongly discourage the use of Asterisk purely as a transit
element for billing. Just because a2billing is available does not mean
you should. Far more scalable solutions are easily available.
--
Sent
Just a laypersons opinion - I'm sure others here have better answers or
justifications.
1. no (at least not realistically, mathematically there are some)
2. perhaps - bandwidth would be your primary concern since 5K calls
would take 150 M of bandwidth
3. IMO it would be better to
2009/2/17 Danny Nicholas da...@debsinc.com
Just a laypersons opinion – I'm sure others here have better answers or
justifications.
1. no (at least not realistically, mathematically there are some)
2. perhaps – bandwidth would be your primary concern since 5K calls
would take 150 M
On Tue, Feb 17, 2009 at 08:57:51AM -0600, Danny Nicholas wrote:
Ok isn't this replacing a western hack with a bridge hack? The init
0 and init 6 probably aren't going to work anyway since (1) asterisk has
to be running as root and
I have already mentioned that this is a requirement.
(2)
Rajkumar S wrote:
Hi,
There are 3 new settings (setinterfacevar, setqueueentryvar,
setqueuevar) and membermacro settings in 1.6 queues.conf. What is
the potential use of these settings? The variables set are useful, but
there is no indication of the purpose they could be used? Any one
Title: Stylish Vacation
You have told us you would like to receive exciting email offers from us.
The sun is getting hotter, the days are getting longer, and summer
Some providers will give it to you on a PRI line. If you are using a
TF number you'll get it regardless.
However keep in mind that it takes me about 3 seconds to change
outbound callerid.
On Mon, Feb 16, 2009 at 9:10 PM, Alfred Monticello ajmce...@yahoo.com wrote:
I'm thinking of starting a
Michael Smith msmith at cbnco.com writes:
Wilton Helm whelm at compuserve.com writes:
There is no reason why it isn't possible to backup in the recorded message
and erase the blip.
Yes, that might be the way to go. I'm playing around with a modified
__ast_play_and_record() that stops
On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:
Hi there,
is gizmo the first user of the Digium Skype solution, or do they use a
different approach/product - any clue?
http://www.gizmo5.com/pc/opensky/
Philipp
OpenSky is no related to any product from Digium. It is a
Hi All,
I upgraded a PBX from 1.2. to 1.4.21.1 and I'm noticing that the hints for
SIP channels are not updating the phones 100% of the time. The hints seem
to work for some time, then the notification on the phone will hang in
either and on or off state. During this condition, on the PBX, core
Hi all,
I just installed Cepstral and app_swift version 1.4.2 on my Asterisk
1.4.22.1 box. It seems to work great with one exception.
If I play a test message with instructions to collect a maximum of 5
digits, it collects those 5 digits correctly if the user waits for the
message to complete
That's funny. The way I have it phrased, when I called I started talking to it
as well! I have some code for short list voice recognition and thought about
detecting yes and no in there, but I ran out of time...and the prompts were
already recorded.
Thank you everyone for helping test the
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 j...@jeff.net wrote:
On Tue, 17 Feb 2009, Andrew Joakimsen wrote:
On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere j...@jeff.net wrote:
Anyone have much
Hey List,
Anyone know the correct way to override an announcement on a queue by queue
basis?
My goal is to have one of my queues say press one to blah.. and no
position announcements I have the jump from queue context working (the
press 1) I just need the correct message played to the user
Rajkumar S schrieb:
How can I stress test an asterisk IVR? I am looking for some kind of
sip phone which can be programmed to send out digits after specified
time to simulate users pressing menu items.
You could remotely control a Snom 3xx like that.
But I guess that's not what you're looking
Christopher Aloi wrote:
Hey List,
Anyone know the correct way to override an announcement on a queue by
queue basis?
My goal is to have one of my queues say press one to blah.. and no
position announcements I have the jump from queue context working (the
press 1) I just need the
Hi,
How do you configure a Patton smartnode to register with an Asterisk server
?
I could do it with 4.2 web server but I'm lots with 5.3 web interface ?
Alternatively, has anyone a correct running-config for that ?
Regards
___
-- Bandwidth and
Mark Michelson wrote:
Christopher Aloi wrote:
Hey List,
Anyone know the correct way to override an announcement on a queue by
queue basis?
My goal is to have one of my queues say press one to blah.. and no
position announcements I have the jump from queue context working (the
press
Asterisk supports SIP-T?
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello Danny,
Thank you for the swift reply!
As it turns out, this was an artifact from ntop, which has a default
maximum buffer size of 8232 bytes.
We're still getting choppy audio, but we've ruled this error message out
as a possible cause.
Thanks again,
Matt.
From: Danny Nicholas
Here's the version -
Asterisk SVN-branch-1.4-r143404
Just static queues.
Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/ dir
for these queue announce files? So my custom file should live in that dir
right?
Thanks for the help :)
On Tue, Feb 17, 2009 at 1:05 PM,
found out that the best solution is to use OpenSips as SIP
OpenSIPS is a great free software proxy.
1- Is there any Software limitation on asterisk regarding number of
simulltaneous calls?
There isn't any explicit limitation in Asterisk or OpenSIPS that I'm aware of,
but you are limited to
Certainly a sobering thought. Have others had to deal with this in PBX
replacement scenarios? Its a giant cost savings in this case - they are
dropping about 12 POTS lines in favor of utilizing (an underutilized) T1
trunk that was already in place.
j
On Tue, 17 Feb 2009, Don E. Wisdom
Christopher Aloi wrote:
Here's the version -
Asterisk SVN-branch-1.4-r143404
Just static queues.
Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/
dir for these queue announce files? So my custom file should live in
that dir right?
Thanks for the help :)
Yah - Found my problem, I can't spell -
periodic-*annouce* = SD-PLS-HOLD
periodic-announce-frequency=10
: )
On Tue, Feb 17, 2009 at 1:19 PM, Christopher Aloi chris.a...@gmail.comwrote:
Here's the version -
Asterisk SVN-branch-1.4-r143404
Just static queues.
Is it true that Asterisk
Christopher Aloi wrote:
Yah - Found my problem, I can't spell -
periodic-*annouce* = SD-PLS-HOLD
periodic-announce-frequency=10
: )
Oh, Ha! That'll do it every time.
Mark Michelson
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snip
Certainly a sobering thought. Have others had to deal with this in PBX
replacement scenarios? Its a giant cost savings in this case - they are
dropping about 12 POTS lines in favor of utilizing (an underutilized) T1
trunk that was already in place.
/snip
Yes -- our alarm monitoring company
Can live in this directory or any under it. If you specify file * looks
in VLAS, if you specify foo/file * looks in VLAS/foo.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Aloi
Sent: Tuesday, February 17,
For those who testing the gender detection module via the number provided:
How was the experience, aside from the funny beep?
In your perception, how well did it perform? (I see raw numbers here, but
perception is important too.)
Do you have any comments, suggestions, or feedback?
After helping out it seems I've been determined a female(wrongly). It
was disappointing and I'm considering a visit to the Dr Phil Show to
work out my anger
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Asterisk
Sent:
David Gibbons wrote:
snip
Certainly a sobering thought. Have others had to deal with this in PBX
replacement scenarios? Its a giant cost savings in this case - they are
dropping about 12 POTS lines in favor of utilizing (an underutilized) T1
trunk that was already in place.
/snip
Yes --
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 POTS line
snip
We will be testing the ADT connection heavily this week. The modem
connections to my understanding are 2400 baud. Over G.711U and a T1 I
don't see why this wouldn't be as solid as a POTS line, but our tests will
tell!
/snip
We do *fax* in this way and it works like a charm. We can hit much
You may be able to split up some of the servers into multiple VMs -- maybe
five servers with five VMs each.
I'm not sure I see the merit in this. VMs seem to be regarded as a magic
bullet (i.e. free lunch). I don't know of any case where 5 VMs can accomplish
more work on one processor than
On Feb 17, 2009, at 1:20 PM, David Gibbons wrote:
snip
We will be testing the ADT connection heavily this week. The modem
connections to my understanding are 2400 baud. Over G.711U and a T1 I
don't see why this wouldn't be as solid as a POTS line, but our
tests will
tell!
/snip
We
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
solid
I have a problem of using call file to make an auto dial out call through
FXO channel. I defined the channel in the call file as Channel:
DAHDI/1/8775203463 When I put the call file under the
/var/spool/asterisk/outgoing dir it did not call out but came to the
context I defined in extensions.conf
On Tue, Feb 17, 2009 at 15:09, Jeff LaCoursiere j...@jeff.net wrote:
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
Jeff LaCoursiere wrote:
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
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 that could be made
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
You should post the call file. Also, I'd use DAHDI/G1 instead of DAHDI/1 as
that ties the call to a specific port/line (perhaps what you want to do?)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ray Chen
Sent: Tuesday,
Accuracy should be 10%-15% better on Wed or Thu.
From: Jason Aarons (US) jason.aar...@us.didata.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 17, 2009 10:48:07 AM
Subject: Re:
is there anything i can do in my dialplan to only ring phones which are not
on a call at the time someone dials in?
its for a call center, they do not want to use queues, but they are
complaining that the call waiting beep is annoying.
i tried call-limit in the sip.conf but then it just busy
National fire protection association
They write the fire codes.
http://www.nfpa.org
On 2/17/09 1:28 PM, Jeff LaCoursiere j...@jeff.net wrote:
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
I will be releasing updated versions to many of the detection modules next
week. They include better support of Asterisk 1.2, 1.4, and 1.6, better
detection, better parameters, an easier build system, and usability is enhanced.
The updated modules include:
* FaxDetect, LineDetect, and
Jeff LaCoursiere wrote:
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
http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650
;p=1
I can't see the Dept Transportation running copper to all the motorist
aid boxes along the highway. I thought most of your alarm panels have
moved to GSM/CDMA backup communications. I'd like to see a fire
marshall
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
system to meet all applicable codes? If (god forbid) someone was hurt in
such a situation and the alarm didn't get passed because of being
Jason Aarons (US) wrote:
In general in the terminology for this stuff supervised just means the
system its referring to not only knows when something bad is happening,
it also is constantly told everything is ok, and timing out waiting for
that ok is also an indication of a problem.
There is
Hello,
Is here any dial plan variable which could help me to identify that call was
dropped (when still not connected) by caller?
HANGUPCAUSE returns 0
DIALSTATUS returns NOANSWER
How to identify such situation?
Related question - how to know which end (caller or callee) ended the
On Tue, 17 Feb 2009, Jon Weisman wrote:
is there anything i can do in my dialplan to only ring phones which are not
on a call at the time someone dials in?
its for a call center, they do not want to use queues, but they are
complaining that the call waiting beep is annoying.
i tried
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
system to meet all applicable codes? If (god forbid) someone was hurt in
such a situation and the alarm didn't get
I think the BAT SIGNAL is the answer.
POTS lines have their issues as well - how many times did we redial to get
into our ISP's in the mid nineties? I have trouble believing the fire
code actually spells out that dedicated POTS lines must be used.
Regradless I think another hold harmless
On 2/17/09 2:05 PM, Jon Pounder j...@inline.net 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
system to meet all applicable codes? If (god forbid) someone was
its about 400 phones, and i dont have access to the tftp server. i was just
looking for a faster way.
thanks,
jon
- Original Message -
From: Gordon Henderson gordon+aster...@drogon.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
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 that
Don E. Wisdom wrote:
On 2/17/09 2:05 PM, Jon Pounder j...@inline.net 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
system to meet
Jeff LaCoursiere wrote:
I think the BAT SIGNAL is the answer.
POTS lines have their issues as well - how many times did we redial to get
into our ISP's in the mid nineties? I have trouble believing the fire
code actually spells out that dedicated POTS lines must be used.
its the
You could set up a hint for the extension and check the hint for inuse
before executing the Dial in your dialplan
Exten = 801,hint,SIP/100
Exten = XXX,1,System(/usr/sbin/asterisk -rx core show hints|/bin/grep
SIP/100|/bin/grep InUse
Exten = XXX,2,GOTOIF($[{CMSTATUS} = FAILURE])?dial
Exten =
It looks like something has changed in the HPET kernel code in 2.6.28
(maybe .27 too) that's stopped ztdummy.c compiling (in 1.2 and 1.4
versions of zapata) A kernel structure member has been renamed with some
crypic comments in the lkml about it.
Anyone know the right thing to do - I'm not
I understand that the Asterisk SLA implementation is somewhat different
from most key systems and PBX systems. I also understand that in
Asterisk, one does not put an SLA line on hold since it is just a MeetMe
conference. However, is there any way to make the BLF flash when the
answering party on
Jon Pounder wrote:
Don E. Wisdom wrote:
On 2/17/09 2:05 PM, Jon Pounder j...@inline.net 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
On Tue, Feb 17, 2009 at 5:00 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Meanwhile what's the 2nd best way to use ztdummy - force it to use the RTC
instead?
The best way to use ztdummy is to read about the change to using
DAHDI, and use dahdi_dummy instead.
Gordon Henderson wrote:
Anyone know the right thing to do - I'm not up on the linux kernel guts,
nor how ztdummy might interact with it, so simply renaming the structure
member (from expires to _expires) is probably not the right thing to do...
If you're already making system changes and
On Tue, Feb 17, 2009 at 1:51 AM, Rajkumar S rajkum...@gmail.com wrote:
How can I stress test an asterisk IVR? I am looking for some kind of
sip phone which can be programmed to send out digits after specified
time to simulate users pressing menu items. If it can originate large
number of calls
In Florida some new subdivision developers have sold the
phone/cable/internet rights to a provider. They run fiber to each house
and then have the uplink to provider which isn't a traditional telco.
You can't get another provider as satellite dishes are limited in
covenants and restrictions (CCR).
On Tue, 17 Feb 2009, David Backeberg wrote:
On Tue, Feb 17, 2009 at 5:00 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Meanwhile what's the 2nd best way to use ztdummy - force it to use the RTC
instead?
The best way to use ztdummy is to read about the change to using
DAHDI, and
Jonn Taylor wrote:
Jon Pounder wrote:
Don E. Wisdom wrote:
On 2/17/09 2:05 PM, Jon Pounder j...@inline.net 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
Muiz Motani wrote:
I understand that the Asterisk SLA implementation is somewhat different
from most key systems and PBX systems. I also understand that in
Asterisk, one does not put an SLA line on hold since it is just a MeetMe
conference. However, is there any way to make the BLF flash when
Here's an alternative to TFTP that works with Polycom 501's. Enable HTTP in
*. Under your static-http directory make a phones dir and put your files
there. In the phone setup, select HTTP and point to
http://1.2.3.4:8088/asterisk/static-http/phones changing 1.2.3.4 to your
local * IP.
John Novack wrote:
Jonn Taylor wrote:
Jon Pounder wrote:
Don E. Wisdom wrote:
On 2/17/09 2:05 PM, Jon Pounder j...@inline.net wrote:
Jeff LaCoursiere wrote:
What do you suppose we have as liability if we are asked to
install such
systems? Is it the
The dial tone for the phone line still comes from the CO. The phone companies
loop there copper cable in and out of the remote cabinets.
Remote terminals are served by T1 or higher density carrier circuits, which can
be either copper or fiber, often employing statistical multiplexing. While
On Tue, Feb 17, 2009 at 1:06 PM, Daviramos Roussenq Fortunato
daviramo...@gmail.com wrote:
Asterisk supports SIP-T?
Nope. Here is some old discussion on this topic:
http://lists.digium.com/pipermail/asterisk-biz/2008-May/026690.html
--
Raj Jain
___
I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on
Sunday in Los Angeles, and the topic of my talk is Open Source in an
Economic Downturn. I've got lots of talking points for this talk,
but it would be interesting to hear some short anecdotes about how you
in the
On Wed, 18 Feb 2009 13:37:57 John Todd wrote:
I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on
Sunday in Los Angeles, and the topic of my talk is Open Source in an
Economic Downturn. I've got lots of talking points for this talk,
but it would be interesting to hear some
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