I'm in southern California, are you close or can you ship?
Bob Knight wrote:
I have 4 sparc based sun boxes I am about to pay money so I can
get rid of them. They are running older versions of Solaris.
You should be able to load Solaris 10 and play around with *
on them.
Time to clean the offi
I have read in wiki pages that for astreisk 1.2.9.1 , you don`t have to install this rpm package.
But Ialso read that Red hat Linux 9 and enterprise doesn`t suppport mp3 sound and song etc.
What are your views??
Regards,
Amna Saleem
On 6/17/06, Sharon Lim <[EMAIL PROTECTED]> wrote:
Did you insta
> On Sat, Jun 17, 2006 at 02:33:26PM -0300, Hermann Wecke wrote:
> > I'm trying to use a Teles (netjet) ISDN BRI card with asterisk
1.2.9.1
>
> It could work with the deprecated chan_modem. Don't wast your time.
>
> > Anyone was able to use this card with asterisk? I couldn't find much
> > inform
>
> I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1
>
> Anyone was able to use this card with asterisk? I couldn't find much
> information about it. Any help?
There is an mISDN driver available on sourceforge:
http://sourceforge.net/projects/misdn4oz
It works pretty well
I'm using Dial(Zap/X/) as suggested.
However, Dial(Zap/X) does indeed work for me. So I'm curious, what's the
difference between them, and when wouldn't just Zap/X work?
On Wed, 2006-06-14 at 11:14 -0500, Eric "ManxPower" Wieling wrote:
> Carey O'Shea wrote:
> > I swear Dial(Zap/X) was the firs
On Sat, Jun 17, 2006 at 09:31:54PM -0500, Aaron Daniel wrote:
> On Sat, 17 Jun 2006, Douglas Garstang wrote:
>
> >Other applications can handle it. Don't see why Asterisk can't. Mount the
> >nfs volume with the -soft option. Do a 'df -k' and you will see that the
> >df command will time out in a
We have an unframed E1 used for data, and it is fiber all the way to our
server room, and then broken out to a G.703 interface.
A few of the E1's I've seen lately for voice have actually been g.shdsl
to the premises with an interface converter between that and the pbx.
You can always rely on your
Hi,
What is the Location. I'm studying in India. Is it possible.
thanks
ArunOn 6/17/06, Bob Knight <[EMAIL PROTECTED]> wrote:
I have 4 sparc based sun boxes I am about to pay money so I canget rid of them. They are running older versions of Solaris.You should be able to load Solaris 10 and play
Thanks Doug, but this would not have helped me. Fortunately the
${INVALID_EXTEN} response was exactly what I needed, but your
suggestion would not have worked, because if an extension is found, it
no longer goes to exten => i. So the variable setting approach ends
up altering the flow of the dia
[EMAIL PROTECTED] wrote:
Where's the Fiber?
I was reading about T1 lines and came across this statement.. It
basically said T1's are made up of copper...Wasn't T1 made up of
Fiber? Is the new trend to move T1 away from fiber and use copper?
Commerical T1 systems were introduced in about 1960
What does your dial command look like?
On 6/17/06, Il Neofita <[EMAIL PROTECTED]> wrote:
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if
I call the traffic still go throw the asterisk. How come?
___
--Bandwidth and Colocati
On Sat, 17 Jun 2006, Douglas Garstang wrote:
Other applications can handle it. Don't see why Asterisk can't. Mount the nfs
volume with the -soft option. Do a 'df -k' and you will see that the df command
will time out in a couple of seconds. Why can't Asterisk do the same?
Just gonna throw ga
The Grandstream seem to be a crap shoot. Some people have real good luck, others don't. So far, I've got four of them in use and the users seem to be happy. The only drawback that I have is that there is no way I can even attempt to try to explain the complex method that you have to use to PARK
I can't tell you how many times I've seen broad questions like this posted to the list..
The wiki (www.voip-info.org) is your friend. Use it. There's a lot of good advise there.
Google is also your friend. Use it, too. Most especially use it to search the list archives. There was just a lo
Il Neofita wrote:
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however,
if I call the traffic still go throw the asterisk. How come?
Are you using the same codecs on the SPA3000 and the xlite? If no
then there's your reason.
--
Linux Home Automation Neil Cherry
I am looking to replace all of the old "Bell" (POTS) phones in my home
and office with IP phones. As you can imagine I don't have a huge
budget to work with but I want phones that will provide acceptable voice
quality and durability.
There are basically three categories as I see it
1. satelli
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come?
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To UNSUBSCRIBE or update options
Other applications can handle it. Don't see why Asterisk can't. Mount the nfs
volume with the -soft option. Do a 'df -k' and you will see that the df command
will time out in a couple of seconds. Why can't Asterisk do the same?
-Original Message-
From: Ira [mailto:[EMAIL
A T1 or E1 or PRI are all made up of copper, at least the last mile
is; that is the last portion between the Telco Demark and the CPE. A
T1 or E1 or PRI can be multiplexed into higher capacity circuits such
as DS3, or OCx and run over transports like ATM and/or SONET. These
higher capacit
Where's the Fiber?
I was reading about T1 lines and came across this statement.. It basically
said T1's are made up of copper...Wasn't T1 made up of Fiber? Is the new
trend to move T1 away from fiber and use copper?
"
http://www.pulsewan.com/data101/pdfs/t1basics.pdf#search='t1%20via%20copper'
At 03:44 PM 6/16/2006, you wrote:
> The "hanging waiting for NFS volume to become avaiable" is a
> classic NFS
> situation, hardly limited to your little experiment.
Silly question, but how is this different than a hard disk in the
local machine crashing or the router dying or even pulling th
Where are they locater ?DovidBob Knight <[EMAIL PROTECTED]> wrote: I have 4 sparc based sun boxes I am about to pay money so I canget rid of them. They are running older versions of Solaris.You should be able to load Solaris 10 and play around with *on them.Time to clean the office:3 Ultra 51 Spar
Im trying to use a MeetMe room to record a podcast. The
quality of the wav is rather poor although all parties entering the room are
coming in via PRI channels. It sounds fine while listening in the room but the
recording is very poor.
Im using lame –h in.wav out.mp3 to convert it. Any
Yes, I have limited access to one SPA-2000 at the moment.
Anyone else seeing this?
When you say open a bug on this do you mean with Asterisk or Sipura?
I guess that's part of the problem, not sure if we should be
troubleshooting on Asterisk or the Sipura device..
On 6/17/06, Rich Adamson <[E
Martin Joseph wrote:
On Jun 17, 2006, at 9:11 AM, Rich Adamson wrote:
I know that the zaptel modules have echo cancellation, but is this
possible to do this on VoIP <-> VoIP traffic as well? I'm toying with
a SIP gateway which has apparently a terrible call quality and would
like to know if
voiplist wrote:
We have issues with all of the SPA-2000 ATAs we have where incoming
calls from only one of our Asterisk servers do not complete.
Details:
1- On the CLI we see that when the call is pushed to the ATA it shows
Busy/Congested
2- We can make calls to this same server just fine
3- We
Thanks Steve, it is good to know that going to the digium card is likely
to reduce the echo.
As you say, it is much cheaper than Sagoma. Here in Canada the Digium
FXO board is just under $100 but a suitable Sagoma with FXO, FXS and
hardware echo can is about $650 !
Mike
Steve Jones wrote:
Can't be done using the 7960 with SIP, unless you are talking about just monitoring that phone. You can monitor a 7960, but you can't show the status of other phones on a 7960 with SIP.
On 6/17/06, Tim Connolly <[EMAIL PROTECTED]> wrote:
Can someone provide an example of how to use HINT pri
Can someone provide an example of how to use HINT priority with
Cisco 7960/SIP phones? I don't fully understand what exactly the hint does,
but I believe it mimics a legacy PBX's bridge-appearance function. Is this
correct?
___
--Bandwidth and
We have issues with all of the SPA-2000 ATAs we have where incoming
calls from only one of our Asterisk servers do not complete.
Details:
1- On the CLI we see that when the call is pushed to the ATA it shows
Busy/Congested
2- We can make calls to this same server just fine
3- We can receive call
Hello, list!
I'm having some trouble with [EMAIL PROTECTED] 2.7(?), Asterisk 1.2.5, inasmuch as
my custom extension is not continuing execution when the caller hangs
up. (Please excuse the sterilized output.)
Here's how it's supposed to go:
exten => 2,8,Monitor(wav,${TIMESTAMP})
exten => 2,9
I'm using a stand-alone VM server and exporting the VM files ro for
MWI function only. All my registration servers mount the remote NFS
share just to check MWI, all read-write functions to the VM files
occur on the VM server only.
On the registration servers, I mounted the remote VM share with t
sjphone
firefly (3rd party version)
--source is available ? where?
On 6/15/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Thu, Jun 15, 2006 at 06:06:40AM -0700, Derek Whitten wrote:
> Asterisk guy wrote:
> > are there any open source sip softphone (Window OS version )?
> > __
On Sat, Jun 17, 2006 at 02:33:26PM -0300, Hermann Wecke wrote:
> I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1
It could work with the deprecated chan_modem. Don't wast your time.
> Anyone was able to use this card with asterisk? I couldn't find much
> information about
On Sat, 17 Jun 2006, Douglas Garstang wrote:
Good grief I hate Outlook webmail. I can't reply inline.
Switch to thunderbird ;)
Anyway, I disagree that all state info except hinting can be replicated. What
about call transfers? If a call is sitting on pbx1, and the user transfers a
call, if
On Jun 17, 2006, at 9:11 AM, Rich Adamson wrote:
I know that the zaptel modules have echo cancellation, but is this
possible to do this on VoIP <-> VoIP traffic as well? I'm toying with
a SIP gateway which has apparently a terrible call quality and would
like to know if there is any way aster
Maybe of you guys know the answer to this:
We have T1's that come from both MCI and Global Crossing as channelized (24
Ports per T) with inband (DTMF) delivery
of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4,
AMI, SF and Wink Start and so is Asterisk. I've moved these T's
Does anyone on-list have an example of working SIP config using Nuvio? I have a
byod account that I'd like to run into my Asterisk server.
Thanks,
Michael
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To UNSU
Maybe of you guys know the answer to this:
We have T1's that come from both MCI and Global Crossing as channelized (24
Ports per T) with inband (DTMF) delivery
of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4,
AMI, SF and Wink Start and so is Asterisk. I've moved these T
>Are you sure that a ro mounted volume won't behave in the same fashion as a
rw mounted one when the NFS server is >abruptly shut down?
>Have you tried shutting down the NFS server? Does Asterisk recover from
this?
Doug,
I have to say, my system has been working fine, but could not recall if I
a
Whare are they located?On 6/17/06, Bob Knight <[EMAIL PROTECTED]> wrote:
I have 4 sparc based sun boxes I am about to pay money so I canget rid of them. They are running older versions of Solaris.You should be able to load Solaris 10 and play around with *on them.Time to clean the office:
3 Ultra
HDSL can sometimes deliver service where copper pairs are nearly exhausted. In other words, if you're down to your last pair of copper, a normal two-pair T1 cannot be delivered, whereas T1 via HDSL can.
On 6/17/06, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote:
[EMAIL PROTECTED] a écrit :> Thanks fo
I have 4 sparc based sun boxes I am about to pay money so I can
get rid of them. They are running older versions of Solaris.
You should be able to load Solaris 10 and play around with *
on them.
Time to clean the office:
3 Ultra 5
1 Sparcstation 5
I also have a box full of Sun keyboards and mi
I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1
Anyone was able to use this card with asterisk? I couldn't find much
information about it. Any help?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mail
JR Richardson wrote:
Pococurante! Or Pococurante? Or you're a big fat poco!
Damn Brian, I had to look this word up.
SYLLABICATION: po·co·cu·ran·te
Pronunciation: (pō"kō-koo-ran'tē, -rän'-, -kyoo-; It.pô"kô-kOO-rän'te)
ADJECTIVE: Indifferent; apathetic; nonchalant.
NOUN: One who does not care; a
On Fri, Jun 16, 2006 at 09:20:18AM -0600, Douglas Garstang wrote:
> Oh... Thanks... This doesn't seem to be documented anywhere.
> Where did you find out about this?
The output of "dump agihtml" gives some hint
"Understands complex variable names and builtin variables, unlike GET VARIABLE."
--
I don't have any links, but there has been work done to 'measure' MOS scores and I believe they are a little more sophisticated than simply tracking latency, jitter and packet loss. An example of one box that does measure/predict MOS is Edgewater Network's Edgemarc. (I have no experience with it so
Good grief I hate Outlook webmail. I can't reply inline.
Anyway, I disagree that all state info except hinting can be replicated. What
about call transfers? If a call is sitting on pbx1, and the user transfers a
call, if it goes to pbx2, Asterisk will complain that it cannot transfer the
call
trixter aka Bret McDanel wrote:
On Sat, 2006-06-17 at 23:25 +0800, Steve Underwood wrote:
Calling MOS totally subjective is rather strange. Telephony only has to
meet subjective goals. In reality, MOS is pretty objective, as it is a
carefully controlled experiment across enough subjective i
JR,
Are you sure that a ro mounted volume won't behave in the same fashion as a rw
mounted one when the NFS server is abruptly shut down?
Have you tried shutting down the NFS server? Does Asterisk recover from this?
Doug.
-Original Message-
From: JR Richardson [mailto:[EMAIL PROTE
I know that the zaptel modules have echo cancellation, but is this
possible to do this on VoIP <-> VoIP traffic as well? I'm toying with a
SIP gateway which has apparently a terrible call quality and would like
to know if there is any way asterisk can help with this.
I believe the current Trun
Jean-Michel Hiver wrote:
Hi List,
I know that the zaptel modules have echo cancellation, but is this
possible to do this on VoIP <-> VoIP traffic as well? I'm toying with a
SIP gateway which has apparently a terrible call quality and would like
to know if there is any way asterisk can help wi
Servetas, Andrew wrote:
I started with Inband, then went to rfc2833 for a while and noticed
other issues with IVR's, so now I'm back to Inband.
http://www.google.com/search?hl=en&q=site%3Alists.digium.com+talkoff&btnG=Google+Search
--
Now accepting new clients in Birmingham, Atlanta, Huntsvil
Yes, we'd need it on every single box. We had a dedicated voicemail server in
the first place. I decided to distribute voicemail between all boxes because
the script that I had that copied the phone registrations over to the voicemail
server (for mwi) was unreliable.
-Original Mess
They exist, but current ones cost a fortune. Used with understanding
they do a fine job. Sadly people who don't understand them tend to read
far to much into the answers they give. Sometimes they are seriously out
of line with perceived quality, but they usually do well.
Yeah I wasnt clear
I have some experience with fibre-channel. I wouldn't be surprised if Asterisk
behaved in exactly the same way if a fibre-channel volume went offline. It's
also prohibitively expensive.
-Original Message-
From: Avi Miller [mailto:[EMAIL PROTECTED]
Sent: Sat 6/17
Hi List,
I know that the zaptel modules have echo cancellation, but is this
possible to do this on VoIP <-> VoIP traffic as well? I'm toying with a
SIP gateway which has apparently a terrible call quality and would like
to know if there is any way asterisk can help with this.
Cheers,
Jean-Mi
On Sat, 2006-06-17 at 23:25 +0800, Steve Underwood wrote:
> Calling MOS totally subjective is rather strange. Telephony only has to
> meet subjective goals. In reality, MOS is pretty objective, as it is a
> carefully controlled experiment across enough subjective individuals to
> filter out a re
I went through the same thing on my home system a couple months ago, and
asked similar questions.. The "conventional wisdom" is that "it
depends"... It depends on your local loop length, quality, taps on the
line, etc..
It seems that most people who have the sangoma cards with hardware echo
ca
trixter aka Bret McDanel wrote:
On Sat, 2006-06-17 at 10:16 +0200, Florian Overkamp wrote:
There are ways to guesstimate MOS scores on a call by continuously
getting some decent statistics from the jitterbuffer. We've had an
intern do some work on this using IAXclient.
http://www.speakup.
Have they released an OpenWengo binary yet that is configuarble and
isn't hardcoded right into their service? You had to build from source
to get the configurable version. Trying to build it from source on
Windows was a nightmare. And once I finally got it to compile, it
crashed on startup. Th
Pococurante! Or Pococurante? Or you're a big fat poco!
Damn Brian, I had to look this word up.
SYLLABICATION: po·co·cu·ran·te
Pronunciation: (pō"kō-koo-ran'tē, -rän'-, -kyoo-; It.pô"kô-kOO-rän'te)
ADJECTIVE: Indifferent; apathetic; nonchalant.
NOUN: One who does not care; a careless or indifferen
On 17 Jun 2006, at 13:58, trixter aka Bret McDanel wrote:
On Sat, 2006-06-17 at 12:52 +0200, Florian Overkamp wrote:
The work that you have done so far is a great step towards a product
that many people might find useful. In a nutshell the concept I am
thinking about is a tool that you drop o
Hi,
I have installed a quadBri card, with
Asterisk-1.0.10 and the bristuff-0.2.0-RC8s (* 1.0.10)
When calling 0207654321 the following
happens:
-- Executing Goto("Zap/1-1 ", "
salsa-helpdesk-day|s|1 ") in new stack-- Goto
(salsa-helpdesk-day,s,1)-- Executing Dial ("Zap/1-1 ",
"Zap/g
On Sat, 2006-06-17 at 12:52 +0200, Florian Overkamp wrote:
> > The work that you have done so far is a great step towards a product
> > that many people might find useful. In a nutshell the concept I am
> > thinking about is a tool that you drop onto your network and it will
> > monitor the data (
Create a context for your queue and put a '*' extension to redirect them
back to the main menu (or wherever)
Also, you're not passing option 'H' to Queue(), right?
'H' -- allow caller to hang up by hitting *.
(I believe the actual hangup digit is defined by features.conf, but I could
be wr
Hi Khairul
We are using 1.2.9.1 and polycom 1.6.6 firmware for
Polycom 601, the problem still exists. It only works for a while and then the
problem happens. But when we reboot the phone, it will work for another while.
I saw a thread about this problem in Digium bug report
page befor
Let me preface this by saying that, I realize I should be asking this
in *-bsd, but I posted there last week and heard nothing so I thought
I would post here to see if anyone had any thoughts:
I just compiled all these from source:
asterisk-1.2.9.1
zaptel-bsd (from svn downloaded on jun 8)
l
Hi,
trixter aka Bret McDanel wrote:
yes and I suggested that however, MOS is an opinion, so its totally
subjective and not based on anything 'real'. That was kinda my point
earlier. Personally I think that its better to isolate the network/cpu
issues and correct them to get what a given implem
On Sat, Jun 17, 2006 at 11:14:33AM +0100, Tim Panton wrote:
>
> On 17 Jun 2006, at 07:53, Kristian Kielhofner wrote:
>
> >Tim Panton wrote:
> >>Well, with 16 phones, it might be worth putting a
> >>'satellite' asterisk in their office, have it handle local
> >>transfers, and act as a protocol con
On 17 Jun 2006, at 07:53, Kristian Kielhofner wrote:
Tim Panton wrote:
Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter, talking sip to the
phones and (trunked) IAX2 to the outside world.
An
[EMAIL PROTECTED] a écrit :
Thanks for the inso...
So T1 lines in the United States also use copper lines from the
company to the telephone exchange in some installations?
What's the benefit to the subscriber to this?
I don't think there is any difference. The E1s I've got at home are
bro
On Sat, 2006-06-17 at 10:16 +0200, Florian Overkamp wrote:
> There are ways to guesstimate MOS scores on a call by continuously
> getting some decent statistics from the jitterbuffer. We've had an
> intern do some work on this using IAXclient.
>
> http://www.speakup.nl/en/opensource/jitterbuffer
Thanks for the inso...
So T1 lines in the United States also use copper lines from the company to
the telephone exchange in some installations?
What's the benefit to the subscriber to this?
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To:
Sent: Friday, June 16
hello!
i'm trying to implement a callback feature. to accomplish this, i've
written a python script(callback.agi) that starts another script as a
independent process(with spawnl), without asterisk waiting for the
other script (callback_dead.sh) to finish before it goes to the next
extension. runni
Hi,
trixter aka Bret McDanel wrote:
MOS (Mean Opinion Score) is generally a bunch of people sitting there
listening to audio and rating it 1-5 (there is a newer method that is
"twice as good" becuase it goes 1-10, basically all values are double).
Its their opinion. This generally cant be dont
Would you believe that somehow I got my [] wrong in cdr_odbc.conf: it
was [general] but should be [global]
I knew I was being stupid.
Sorry for the waste of bandwidth :(
Julian
Jean-Michel Hiver wrote:
Julian Lyndon-Smith a écrit :
svn trunk.
I'm trying to get cdr to work with my odbc dat
Julian Lyndon-Smith a écrit :
svn trunk.
I'm trying to get cdr to work with my odbc database. I have followed a
checklist that I had previously but still can't get it to work. There
are no errors (verbose 40 and debug 40), I get
[cdr_odbc.so] => (ODBC CDR Backend)
== Parsing '/etc/asteris
On Fri, Jun 16, 2006 at 09:40:35AM -0600, Mike Diehl wrote:
> I don't know how big your voicemail system is, but have you considered using
> Unison to syncronize the vm accross all your servers? I'm deploying multiple
> servers with two vm servers, each sync'ed every 5? minutes. If one fails,
Hi,
Does anyone know how to change the DTMF twist?
Please reply.
Anil
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svn trunk.
I'm trying to get cdr to work with my odbc database. I have followed a
checklist that I had previously but still can't get it to work. There
are no errors (verbose 40 and debug 40), I get
[cdr_odbc.so] => (ODBC CDR Backend)
== Parsing '/etc/asterisk/cdr_odbc.conf': Found
*CLI> c
We use Unison Doug and it works just fine. It isn't perfect in theory but we've had no issues in practice. Your concerns over sacalbility are resolved by implementation - do you need it on every single Asterisk box, or maybe local to just two with routing to them and failover in the dial-plan? Unis
That sounds fine except where registrations are involved although I'd suggest you look into SRV as well as RR for the DNS to more finely balance the load for clients which support it. Doug's mail says it all where registrations are involved - not all state information is stored in the database so y
Douglas Garstang wrote:
I don't think unison is a workable solution. It doesn't scale. The network and
system load would increase exponentially as we added asterisk servers to our
cluster.
If you're clustering that many boxes, I'd investigate fibre channel SAN
and GFS. That way, each node of
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