Hello,
I need to use the ${DATETIME} macro inside the filename saved by Record,
but the colons (':') used in the time interfere with the command
(everything after the colon is interpreted as the format I wish to save to):
My command is:
2008/5/2 Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, May 02, 2008 at 09:06:01AM +0200, Vinz486 wrote:
2008/4/30 Tzafrir Cohen [EMAIL PROTECTED]:
On Wed, Apr 30, 2008 at 09:07:48PM +0200, Vinz486 wrote:
On Monday 12 May 2008 01:36:04 Daniel Grad wrote:
I need to use the ${DATETIME} macro inside the filename saved by Record,
but the colons (':') used in the time interfere with the command
(everything after the colon is interpreted as the format I wish to save
to):
My command is:
Hello list,
I've done some work with basic parsing of extensions.conf in order to
generate some visualizations of the dialplan. I've just posted it this past
weekend over on the Asterisk-Java blog at asterisk-java.org. There's a Java
web start demo if you have your extensions.conf handy.
Cheers,
Tilghman Lesher wrote:
On Monday 12 May 2008 01:36:04 Daniel Grad wrote:
I need to use the ${DATETIME} macro inside the filename saved by Record,
but the colons (':') used in the time interfere with the command
(everything after the colon is interpreted as the format I wish to save
to):
My
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zhao_x_q wrote:
I have test G.722 for many phones. I have try calls between sip G.722,
sip G.722 to sip G.711, G.722 to RRI cards, PRIcards to G.722. I also
test meetme conference. Other phones such as grandstream and fanwei have
no problems. The sounds is good, grandstreams have little
Dear all, I have installed asterisk 1.4.13 and configured all the
/etc/asterisk files very well. Always I enter the CLI (with asterisk
-r) and when I make a change after that I execute module reload
and everything is OK.
But a few days ago, without make any change, I execute module reload
Has anybody got any scripts for a lone worker system using Asterisk
before I write them?
Something along the lines of a regular phonecall with some kind of
random question (e.g. press 1 then 5) to provide monitoring of lone
workers with alerts?
Steve
The information contained in this
I've been having a look at some of the information on the website of
Voicetronix in Australia, and see that their cards make use of wanpipe
and wanrouter. I already knew that Sangoma cards also make use of
those, so my question is: what is the relationship, if any, between
Voicetronix and Sangoma
Sorry to be a pest, but does anyone have any ideas on this? I've
opened a bug, but I was hoping someone else on the list has
encountered this issue before.
Thanks,
Jason
On May 9, 2008, at 12:36 PM, Jason Dixon wrote:
We have a remote office that's having problems with their Polycom.
On Mon, May 12, 2008 at 10:28 AM, Steve Hanselman [EMAIL PROTECTED] wrote:
Has anybody got any scripts for a lone worker system using Asterisk before I
write them?
Something along the lines of a regular phonecall with some kind of random
question (e.g. press 1 then 5) to provide monitoring
Hello All,
Anyone purchased a asterisk card, x100p or similar in India, if yes from
where and what model ? I am interested in setting up a Asterisk Server at
home, for single line at the moment and if things work out great, I would
like to migrate that to my business and replace the aging pbx
We have been using Sangoma A200 for about an year now with BSNL connection. I
don't know if you can get it in India directly as in our case it was brought
from US directly.
Regards,
Sanjay Rajdev
- Original Message -
From: Amit Patel [EMAIL PROTECTED]
To:
On Mon, May 12, 2008 at 10:36 AM, Tony Mountifield
[EMAIL PROTECTED] wrote:
I've been having a look at some of the information on the website of
Voicetronix in Australia, and see that their cards make use of wanpipe
and wanrouter. I already knew that Sangoma cards also make use of
those, so
He wants a randomly generated phone call to be generated to a specific
extension.
Eg once an hour for the midnight to dawn shift at a random time per
hour.
When the person picks up they are asked a question using an audio file.
(or text to speech).
Then the person has to enter the correct dtmf
Hi,
I read the WiKi, which implied there was a way of working around this,
but the HTML nature of the WiKi seems to have destroyed some of the
output so I cannot see the correct answer...
I would like to match a special case of a number dialled 0x, now
normally I would simply do:
exten
Eric Wieling wrote:
People that try to wing it and install Asterisk when they don't know
telecom just gives people a bad impression of Asterisk and VoIP in
general. This helps nobody except the pocketbook of the consultant.
I agree. But I think that comment is incredibly funny. I'd like to
Spot on (except for the shitty way, it's pretty standard, in building
there are paging systems that start an escalating tone, beyond the
building these don't work, so if you're away then we'd be dialling the
mobile).
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Gentlemen,
First let me say it's great to be back on the Asterisk mailing lists.
Those of you who have been around for a while will remember me as
Rushowr. I look forward to answering questions and whatnot in the
future, but for the moment I have a minor question that I cannot find a
Sorry to be a pest, but does anyone have any ideas on this? I've
opened a bug, but I was hoping someone else on the list has
encountered this issue before.
Jason
Does the Polycom have the Buddy List turned on? We had an IP601 that
would reboot (or lock up) about 60% of the time when IT
So, do you mean that if :
1. Asterisk server boots,
2. A cable from telco analog line is plugged in and out in every FXO port
3. Analog lines (from Telco) are plugged into definitive FXO ports
4. Then, any query to InAlarm field would tell if a cable is plugged or not
?
Hi,
What is the syntax to set more than one variable in the SIP.conf file for a
particular sip peer? (using the setvar line)
Regards,
Mick
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To
2008/5/12 Olivier [EMAIL PROTECTED]:
So, do you mean that if :
1. Asterisk server boots,
2. A cable from telco analog line is plugged in and out in every FXO port
3. Analog lines (from Telco) are plugged into definitive FXO ports
4. Then, any query to InAlarm field would tell if a cable is
Dear all, I have installed asterisk 1.4.13 and configured all the /etc/asterisk
files very well. Always I enter the CLI (with asterisk -r) and when I make
a change after that I execute module reload and everything is OK.
But a few days ago, without make any change, I execute module reload
I have an Asterisk 1.4.19.1 server that is behind a Fortinet firewall.
Localnet is 192.168.2.0/255.255.255.0 and all external sip devices look
as if they are on the same local network because the Fortinet rewrites
the incoming IP as its own address.
The problem I have is that when
srv04*CLI show application Dial
srv04*CLI
-= Info about application 'Dial' =-
[Synopsis]
Place a call and connect to the current channel
*SNIP*
p- This option enables screening mode. This is basically Privacy mode
without memory.
P([x]) - Enable privacy mode. Use 'x' as
Hello All,
I've been having some intermittent trouble with an Asterisk 1.2.10
installation that is supporting roughly 50 SIP clients on a LAN, mostly
soft phones and about 10 snom VoIP phones. We have a custom soft phone
client which displays presence information for various extensions.
Tzafrir Cohen wrote:
On Fri, May 09, 2008 at 11:50:59AM -0400, Drew Gibson wrote:
equis software wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what
do you think about to use Ubuntu or another distibution??
Thanks
I have run Asterisk on several
I'm not sure if a full-height card would fit (vertically) in a 3U chassis...
but I would probably also assume that if it would not, that the chassis/mobo
would have a PCI/PCI-Express riser card that would mount the cards horizontally.
Might want to check that out with the manufacturer of the
Hi all,
on my debian box i configured chan_oss to work with /dev/audio device.
CLI console command and Dial(CONSOLE/dsp) work perfectly but i notice
2 problems:
1. audio is very low in volume, even if i set 100 the mixer volume
(via cmd line setmixer utility)
2. the sound is very crappy: the
Matt Watson wrote:
I'm not sure if a full-height card would fit (vertically) in a 3U chassis...
but I would probably also assume that if it would not, that the chassis/mobo
would have a PCI/PCI-Express riser card that would mount the cards
horizontally.
Might want to check that out with
Hello All,
Is there a way to have Manager Bridge Channel to the specified extension
without the channel being connected.
In the current scenario the channel only bridges once the call get connected,
it does not bridge when any service provider (telco) message is played. I want
to record all
Getting the RIGHT card for the RIGHT bus type and the RIGHT Chassis is
NOT as simple
as everyone will lead you to believe.
My suggestion, worth exactly what you paid for it :)
Get Exact Spec for the card your are considering and FAX / Email to PC
vendor and have him
send you In Writing
I've got the text files created -- thanks to Russell Bryant -- for
re-building the core and extra sounds using another voice but I'm not
sure which formats to actually build.
This will be a small/personal system using Vitelity.net so will only
have SIP connections.
The
On Mon, May 12, 2008 at 3:15 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Matt Watson wrote:
I'm not sure if a full-height card would fit (vertically) in a 3U
chassis... but I would probably also assume that if it would not, that the
chassis/mobo would have a PCI/PCI-Express riser card
Asterisk will automatically chose the best format - per ATFOT
Roderick A. Anderson wrote:
I've got the text files created -- thanks to Russell Bryant -- for
re-building the core and extra sounds using another voice but I'm not
sure which formats to actually build.
This will be a
A quality 3U chassis will mount the cards parallel to the mainboard
with the use of a riser card, just as a 1U chassis does.
If you are intent on sourcing the components yourself may I suggest a
Tyan or Supermicro barebones server? I think that is the best
solution for integration in these sort
Does anybody know if i can make (chan_mobile) module to be installed and work
with Asterisk 1.4.19 ?___
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On Tue, May 13, 2008 at 3:44 AM, gres [EMAIL PROTECTED] wrote:
Does anybody know if i can make (chan_mobile) module to be installed and
work with Asterisk 1.4.19 ?
Make sure you have the dependencies (bluez*) and do a make menuselect
when compiling Asterisk.
Good info and links
P.S. Hope your lone worker is paid a lot to be working for a shitty
company checking up on them like that :)
Actually, in the UK, a company has a duty under the Health and Safety at
Work etc Act of 1974, and the Management of Health and Safety at Work
Regulations of 1999.
'Employers have
Hach Segal a écrit :
Hello All,
I've been having some intermittent trouble with an Asterisk 1.2.10
Before anything else did you tried an updated asterisk 1.2
The last one is 1.2.28 or something like that, and there has been
a lot of security patches, and fixes since your version.
Did you
I need SIP trunking from a high quality business service provider for
25,000 SIP minutes growing at approximately 10% each month.
Currently we are using exgn.net to provide inbound 800 (Costs $200 for
approx 10,000 minutes a month)
and we are using broadvoice.com for outgoing calls (Costs $100
Andreas van dem Helge wrote:
Best is to keep native format files for all codecs you intend to use.
Where are these rebuilt files with another voice? And any chance
they'll ever be done by Pat Fleet or Ann, the Cisco voice? FWIW I'm
in contact with someone who's been in contact with Pat who
Al Baker wrote:
Asterisk will automatically chose the best format - per ATFOT
I guess I'm not getting my head wrapped around this concept. I
understand the choosing but not how I might influence it. Probably best
to just build them all and let Asterisk sort it out. I'll research this
some
I have read the post about the touch tone before to connect so
transfered calls don't end up in voicemail boxes of mobile phones. I
have done some work last year on transfering an inbound call to
different extensions by using meetme() and local channels so a whole
group can start talking.
There are krone blocks designed for CAT5, and I've seen them in use as well.
However, there's no way I'd be using them for today's networks.
/Especially/ having seen one of these krone blocks used to double-punch
two network ports together.
Bill Andersen wrote:
Oh, yes. I saw an entire
Tilghman Lesher wrote:
On Monday 12 May 2008 17:27, Roderick A. Anderson wrote:
Al Baker wrote:
Asterisk will automatically chose the best format - per ATFOT
I guess I'm not getting my head wrapped around this concept. I
understand the choosing but not how I might influence it. Probably
In The future of Telephony, it says ... We should also note for
security's sake you should always make sure that your [incoming] context
never allows outbound dialing. (If by chance it did, people could dial
into your system and make outbound toll calls that would be charged to
you!)
The book
At 9:43 AM on 13 May 2008, Lee, John (Sydney) wrote:
In The future of Telephony, it says ... We should also note for
security's sake you should always make sure that your [incoming]
context never allows outbound dialing. (If by chance it did, people
could dial into your system and make
Additional info.
I tried to disable chan_oss and enable chan_alsa (seems like kernel
2.6 should use alsa and not oss).
Well, no dial CLI command, no Dial(console/dsp) channel available.
So, how to use chan_alsa?
--
PicoStreamer - the real WEB live streaming software
vinz486.com
On Monday 12 May 2008 18:44, Roderick A. Anderson wrote:
Tilghman Lesher wrote:
On Monday 12 May 2008 17:27, Roderick A. Anderson wrote:
Al Baker wrote:
Asterisk will automatically chose the best format - per ATFOT
I guess I'm not getting my head wrapped around this concept. I
With an ISDN10/20/30/etc, I would just put all the lines into an
'incoming' context - and make sure that incoming context doesn't have
any includes (unless you really need them...)
PaulH
On Tue, 2008-05-13 at 09:43 +1000, Lee, John (Sydney) wrote:
In The future of Telephony, it says ... We
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