Hi everybody !
I'm desperately trying to install AsteriskNow Beta 6. I
downloaded the iso file (version x86 32 bits) and burned it,
then I tried on three different computers (from an old Pentium
4 to a brand new HP DL380 2xDual Core) and each time I got the
same error...
Shortly after the
On 7/6/07, C F [EMAIL PROTECTED] wrote:
Have you tried wav49 format?
Yes, I have
format=wav49|wav
Mike
On 7/6/07, Mike Dent [EMAIL PROTECTED] wrote:
Hi,
I recently upgraded the firmware on my Blackberry 8700 to 4.2, this
seems to give
it the ability to play wav files.
I wondered
On 7/6/07, Dave Bour [EMAIL PROTECTED] wrote:
Can you see an attachment? If so, does it download?
Yes the attachment is there but it seem that it will not download it,
which leads me
to believe it does not understand the format?
thanks,
Mike
Dave Bour
Desktop Solution Center
Here's a quick report about this:
During the entire hour, there were always around 25 people watching
the video stream. I've recorded it locally and will put the pertinent
stuff (questions and Mark's answers) up on http://asterisktv.com bit
by bit as soon as I have them edited. The audio
On 4 Jul 2007, at 17:57, Stephen Bosch wrote:
Jaswinder Singh wrote:
Think about voicesense which will sense what you are talking and
pop in
a *relevant* voice ad to spice up conversation :P .
If this happens I am going back to tin cans and string.
Hmm, time to get that IAX encryption
On 5 Jul 2007, at 17:50, Steve Davies wrote:
Hi,
Just a quick question. Is there a way when making an IAX call to
transmit some additional call-data, perhaps in a variable? I could
overload callerid-name, but that is nasty and ugly :)
Yep, there is the 'IAXVAR' patch that does exactly
I just tested on mine (7130...non-media supporting yet), in the message,
it says there's an attachment but the BB itself doesn't register an
attachment (ie, if on the main email screen, no paperclip on the
envelope). It doesn't give an option to down as a result. Do you have
the same?
It shows the attachment in the message but there is no paperclip symbol ?
I forwarded the email to my wifed BB Pearl and it opened and played
just fine. Including showing the paper clip in the message list.
So it would seem my 8700 with 4.2 software is not capable of playing
these vm attachments.
Time to upgrade ;)
D
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]
For those who just want it to work...
Giving you complete IT peace of mind.
(Sent via Blackberry - hence message may be shorter than my usual verbose
responses)
PIN 4cc364db (as of March 24, 2007)
-
Haha, I like the full keyboard on my 8700 too much
Maybe when the 8820 is out I will
Mike
On 7/7/07, Dave Bour [EMAIL PROTECTED] wrote:
Time to upgrade ;)
D
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]
For those who just want it to work...
Giving you complete IT
Reposted to this list: (http://lists.virus.org/voipsec-0610/msg00046.html)
That's exactly the type of thing that needs to be stopped. If Dell
outsourcing calls me from India, the CLI must be their number in India
not a faked-in number of some office in the US. That to me is exactly
the
Yes, I bit at the condensed keyboard on the 7130 but it's been the worst
quality audio and construction of any blackberry I've had
Truth be told, I'm ready to go back to the full keyboard too
D
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]
For those who just want it to
I asked some of the devs about this a while ago - and the answer was no,
you have to set each hint by hand. :(
Ah well.
PaulH
On Thu, 2007-06-28 at 12:06 -0500, Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql
You can use asterisk db functions to set an afterhours flag, and then
use the dialplan to check if the flag is set or not.
One of the techs in our office does stuff like that all the time for
clients.
PaulH
On Tue, 2007-07-03 at 17:00 +1000, Farooq Ahmed wrote:
Hi all,
As we know we can
On Sat, 7 Jul 2007, Olivier wrote:
Hi,
My setup is :
PSTN - ISTP Network --- Router - Asterisk
-- SIP Phones
Phones are located in the same location.
I'm thinking about installing new phones in other locations (small agency,
home workers), registering
On Sat, 2007-07-07 at 08:39 -0500,
[EMAIL PROTECTED] wrote:
Date: Fri, 06 Jul 2007 12:02:53 -0600
From: Stephen Bosch [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office.
To: Asterisk Users Mailing List - Non-Commercial Discussion
On the other hand, the guy could just be using his work e-mail for personal
interests.
On 7/7/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
On Sat, 2007-07-07 at 08:39 -0500,
[EMAIL PROTECTED] wrote:
Date: Fri, 06 Jul 2007 12:02:53 -0600
From: Stephen Bosch [EMAIL PROTECTED]
Subject: Re:
I don't believe AsteriskNow will install on a dual processor system. I
had this same problem - installing on single process MB went OK
I don't know how to fix, so went with elastx.org and adminsparadise.com
packages, both seemed to be OK - can't decide which one to keep - the
last choice, maybe
Just a quick listing of tested, and updated, steps from my notes.
Enjoy !
http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html
-baji.
--
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Carlos:
I think you problem is on the queue systems, we have the same problem
on version 1.2.x and 1.4.x
one one of our call centers.
Try to change you agents to be dynamic, and also to change the login
method from AgentLogin to AgentCallBackLogin
Alvaro
On 6/8/07, Carlos Chavez
Hi Everyone,
I'm planning my first asterisk box, and I'd like to know what SIP
providers everyone likes. Voipjet? Gizmo? Somebody else?
Thanks,
Alex
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
Hi list:
I want to modify the libmfcr2. But i can't find where is define the end
DNIS signal is define. Actually the libmfcr2 send a ONE (1) at the end of
sending all the DNIS numbers. I need to send a TWO (2), this becouse in
Mexico the normal is to send a 2 at the end, not a ONE.
In the
On Sat, 7 Jul 2007, Alex Roston wrote:
Hi Everyone,
I'm planning my first asterisk box, and I'd like to know what SIP
providers everyone likes. Voipjet? Gizmo? Somebody else?
Which country are you in?
Asterisk is being used all over the planet, and there are SIP providers in
nearly every
True. But I think that fuzzy distinction is also relevant to the fuzzy
process. I'm not talking about suing or fighting anyone, with actual
evidence suitable to that kind of action. I'm just talking about clues
for looking for actual evidence of actual actions.
Besides, Mushtaq
On Sat, Jul 07, 2007 at 12:02:41PM -0400, Baji Panchumarti wrote:
Just a quick listing of tested, and updated, steps from my notes.
Enjoy !
http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html
1. If you're recommending Centos4 (and not Centos 5, or Debian),
Tzafrir,
You make good suggestions, and you have raised good questions.
Unfortunately I have yet to install * on CentOS 5, and all I am doing
here is sharing my notes to facilitate other newbies.
However my blog is available for you to share your ideas and expertise
with the community.
I'm in the US. California, specifically.
Thanks,
Alex
Gordon Henderson wrote:
On Sat, 7 Jul 2007, Alex Roston wrote:
Hi Everyone,
I'm planning my first asterisk box, and I'd like to know what SIP
providers everyone likes. Voipjet? Gizmo? Somebody else?
Which country are you
Hi
I am trying to build reliable fax solution with asterisk, iaxmodem and
hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium 3
1.2GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After
installing
the newest zaptel and wanpipe-3.1.0 beta I did zttest and it didn't give
At 09:11 AM 7/7/2007, you wrote:
I'm planning my first asterisk box, and I'd like to know what SIP
providers everyone likes. Voipjet? Gizmo? Somebody else?
Teliax has been the best for me and I use callwithus as a backup.
Ira
___
--Bandwidth and
Andrew Nowrot wrote:
I am trying to build reliable fax solution with asterisk, iaxmodem and
hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium 3
1.2 GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After
installing the newest zaptel and wanpipe-3.1.0 beta I did
On Sat, Jul 07, 2007 at 08:29:57PM +0200, Andrew Nowrot wrote:
Hi
I am trying to build reliable fax solution with asterisk, iaxmodem and
hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium 3
1.2GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After
installing
the
On 7/7/07, Alex Roston wrote:
I'm in the US. California, specifically.
I guess the other question is what is your need ?
Inbound (need a DID), outbound only (don't need a DID)
or both in outbound ?
Teliax charges $4.95 / mo / DID and 2cents /min for in out
if you use pay-as-you-go.
You are definitely not the only one. For me it is the same. I did sent a
message about this to the list owner.
grz,
Hans Feringa
Hello list,
I am getting the list with days of delay, take for example this message:
Received: from unknown (HELO lists.digium.com) (216.207.245.17) by
Hello all,
I have some polycom 430's which I'm trying to get to work with asterisk.
I have them working for the most port other than one little issue.
The 430's have two line appearances. I'm trying to get the second line
registered to a different extension but for some reason it's not
On 7/7/07, Tzafrir Cohen wrote:
On Sat, Jul 07, 2007 at 12:02:41PM -0400, Baji Panchumarti wrote:
Just a quick listing of tested, and updated, steps from my notes.
Enjoy !
http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html
[...]
3. The process
On 7/7/07, Lee Howard [EMAIL PROTECTED] wrote:
HI
Are you having trouble with fax? Rumor is it that the Sangoma hardware
isn't as needy this way as is the Diguim. I'm not sure about that,
though.
I heard that too, but unfortunately I have some problems with incoming
faxes (but only when
Hi
The kernel timer shouldn't be relevant. The timing should come from the
card, and not from ztdummy. Make sure that the timing comes from the
card and not from ztdummy.
I don't load the ztdummy module, so timing is taken only from the card. The
only reason I put rtc in my kernel is that I
On Sat, Jul 07, 2007 at 04:18:38PM -0400, Baji Panchumarti wrote:
On 7/7/07, Tzafrir Cohen wrote:
On Sat, Jul 07, 2007 at 12:02:41PM -0400, Baji Panchumarti wrote:
Just a quick listing of tested, and updated, steps from my notes.
Enjoy !
Ira wrote:
At 09:11 AM 7/7/2007, you wrote:
I'm planning my first asterisk box, and I'd like to know what SIP
providers everyone likes. Voipjet? Gizmo? Somebody else?
Teliax has been the best for me and I use callwithus as a backup.
I'd have to agree that Teliax has been the best for me as
Jeff Davis wrote:
I'd have to agree that Teliax has been the best for me as well. I've
used them for over a year now with only a couple of minor administrative
problems.
All ITSPs suck! (tm)(c) 2007 ManxPower
Teliax seems to suck less than most.
In addition they maintain a very large stock
Hi,
I have (to me) an interesting problem.
There are 3 physical extensions, 11, 12 and 13. All hang off Sipura
adapters.
There is also extension 10 which simply uses 'Dial(SIP/11SIP/12SIP/13)' to
call all phones in the house.
Incoming calls from outside get sent to 10 in order that they can be
On Sat, 30 Jun 2007 21:04:08 +1000, Bill Maidment wrote
Hi guys
I'm at a loss in getting ./configure to complete successfully with asterisk
1.4.6 on Fedora 7 x86_64, as it complains about no termcap support, even
though it is installed
Wow! It took 8 days for my email to register
Eric ManxPower Wieling wrote:
Jeff Davis wrote:
I'd have to agree that Teliax has been the best for me as well. I've
used them for over a year now with only a couple of minor administrative
problems.
All ITSPs suck! (tm)(c) 2007 ManxPower
Teliax seems to suck less than most.
They should
Is it possible to detect SIT tones on an outbound call?
Specifically this is for an outbound call generated via a .call file
over a PRI. I get answer supervision when the SIT tone starts and
Asterisk believes this is a successful call.
I'm using Asterisk 1.4.6.
Thank you!
Hi all,
My scenario is such that I have three users connected to a conference.
CLI meetme list 1234
User #: 01 9176502096 no nameChannel: Zap/23-1
(unmonitored)00:00:32
User #: 02 john john Channel: SIP/john-b7800468
(unmonitored) 00:00:28
User #: 03
Hello Asteriskers,
I'm confused about why Asterisk is not a SIP proxy and why exactly
this can affect the performance of a large Asterisk system.
I know that Asterisk acts as a useragent endpoint, but my doubt is why
exactly Asterisk could overload the call flow if the RTP voice stream
goes from
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