information
983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio,
ITU G.711 A-law, mono 8000 Hz
But when i install the same system on Centos 5.5 (kernel 2.6.18-92.el5
#1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) i
could get the same information with file command
Use Addmember and removemeber instead :)
l.
2010/5/14 Peter Childs pchi...@bcs.org
I've been trying to get the hang of Agents and Queues and I must say
its a little unclear as to how things work.
So maybe someone has some better idea
From what I can work out an Agent is meant to
Vieri wrote:
--- On Fri, 5/14/10, Steve Edwards asterisk@sedwards.com wrote:
I'm supposing my system is using the DAHDI-driven
Digium cards on my
motherboard. I don't know how hardware timers work and
if Digium
hardware rely on the motherboard (my system clock is
going too fast
Hello,
you know , when a call setup, either caller hangup first or callee
hangup first , the hangupcause will set to 16(means Call Clearing
Causes)
My question is how could i identify whether the caller or callee
hangup the phone first?
Best Regards!
--
Thanks for your supporting,
have a nice
Am 17.05.2010 10:46, schrieb Zhang Shukun:
Hello,
you know , when a call setup, either caller hangup first or callee
hangup first , the hangupcause will set to 16(means Call Clearing
Causes)
My question is how could i identify whether the caller or callee
hangup the phone first?
AFAIK
2010/5/14 --[ UxBoD ]-- ux...@splatnix.net:
- Original Message -
Hello,
i try to use soap in the phpagi.
i want to call a function from a web service
when a user call a telephne failed.
this is my phpagi script, Could you show me what's wrong ? becasue i
can't excute it
Hello everyone
---
Regards,
Raj.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
Klaus Darilion wrote:
Am 17.05.2010 10:46, schrieb Zhang Shukun:
Hello,
you know , when a call setup, either caller hangup first or callee
hangup first , the hangupcause will set to 16(means Call Clearing
Causes)
My question is how could i identify whether the caller or callee
hangup
- Original Message -
2010/5/14 --[ UxBoD ]-- ux...@splatnix.net:
- Original Message -
Hello,
i try to use soap in the phpagi.
i want to call a function from a web service
when a user call a telephne failed.
this is my phpagi script, Could you show me what's wrong
I think You can do this if you use local channel.
For example, you do two context
AEL example
context Incoming {
_X. = {
..
Dial(Local/${ext...@outgoing/n);
};
h = {
Noop(Hangup in Incoming);
}
};
context
And the first hangup message is the hangup callee or caller
If you become first message Hangup in Incoming, that mean the hangup was
make in Incoming context
If you become first message Hangup in Outgoing, that mean the hangup was
make in Outgoing context
Vardan
P.S.
[default] // In example
Hello,
I put together a new tutorial about asterisk realtime integration with
kamailio (openser). This time the database used is the one of asterisk,
also call routing logic is controlled by asterisk, here is the link:
http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
Have you looked at Dial's g option ?
*g*: When the called party hangs up, exit to execute more commands in the
current context.
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Yes, I know about this option, You can you this also, but, how you want
to see what leg was hangup the channel?
Olivier wrote:
Have you looked at Dial's g option ?
*g*: When the called party hangs up, exit to execute more commands in
the current context.
--
Vardan Harutyunyan,
Senior
Hi, is it possible to add a context from the console using the dialplan
command?
Thanks
Lee
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kb.asipto.com isn't reachable: DNS doesn't resolve the domain name.
Alex
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Daniel-Constantin
Mierla
Inviato: lunedì 17 maggio 2010 12:01
A:
Same problem here.
---fred
On May 17, 2010, at 6:28 AM, Alexandru Oniciuc wrote:
kb.asipto.com isn't reachable: DNS doesn't resolve the domain name.
Alex
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per
in my case it work
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Alexandru Oniciuc wrote:
kb.asipto.com isn't reachable:
only). Using file command i can get the following
information
983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio,
ITU G.711 A-law, mono 8000 Hz
But when i install the same system on Centos 5.5 (kernel 2.6.18-92.el5
#1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64
On Mon, May 17, 2010 at 12:36 PM, Fred Posner f...@teamforrest.com wrote:
Same problem here.
---fred
On May 17, 2010, at 6:28 AM, Alexandru Oniciuc wrote:
kb.asipto.com isn't reachable: DNS doesn't resolve the domain name.
Alex
I see the DNS resolving on our Virginia server but not here
On 5/17/10 1:02 PM, Randy R wrote:
On Mon, May 17, 2010 at 12:36 PM, Fred Posnerf...@teamforrest.com wrote:
Same problem here.
---fred
On May 17, 2010, at 6:28 AM, Alexandru Oniciuc wrote:
kb.asipto.com isn't reachable: DNS doesn't resolve the domain name.
Alex
I
so you can go for it.
the basic idea:
Dian(***,***,g)
Noop(Called party hung up first)
Hangup
On Mon, May 17, 2010 at 1:23 PM, Vardan hvarda...@gmail.com wrote:
Yes, I know about this option, You can you this also, but, how you want
to see what leg was hangup the channel?
Olivier wrote:
And how you want check the both leg?
Look, you have to channel.
Incoming and Outgoing.
You want to see, who first do hangup.
How you want do this with g?
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of
Works for me
Thanks,
Hristo Benev
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandru Oniciuc
Sent: Monday, May 17, 2010 6:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On 17 May 2010 08:40, Lenz Emilitri lenz.lo...@gmail.com wrote:
Use Addmember and removemeber instead :)
l.
Hmm I'm getting that kind of.
From What I can work out.
Agents have been deprecated and are going to be removed.
The replacement, is some complex dialplan using Local Channels which
Using Asterisk 1.4.31 and addons 1.4.11, ControlPlayback get confused
when skipping forwards on an mp3 file (it seems to work fine on wav's).
I'm calling it from an AGI like so:
$agi-exec('ControlPlayback',$filename . |4000|#|*|8|0|7);
The first four times I press the '#' key it does indeed
Here is what I do to handle agent login/logout
; Agent login logout
exten = *20,1,Verbose(2,Doing agent login/logout)
exten = *20,n,Answer()
exten = *20,n,wait(.0.5)
exten = *20,n,Read(AgentNumber,agent-user)
exten = *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten =
Bruce-
On 05/16/2010 11:22 AM, Jeff Brower wrote:
Bruce-
I'm trying to make an AMI call. I want to call a number, play an
announcement when the call is answered, then call a second number and
connect the two when the second call is answered.
I an able to make a simple call to two
Hi Jim,
First i'm a little bit confused, because your code was a little bit difficult
to read, but now i understand.
it works fine in my Setup, Big Thanks for your help.
bye
Daniel
Am 16.05.2010 um 16:11 schrieb Jim Dickenson:
We do the following:
Action: Originate
Channel:
Hi Kevin,
We don't have mohinterpret set at all, so I think it uses default.
Is there anything else you can suggest? Any other places to go for
help?
Thanks for your assistance!
On Thu, May 13, 2010 at 11:32 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/13/2010 05:16 PM, David
How does Asterisk (1.2) handle a 180 WITH SDP?
I am seeing different behavior when a call is initiated from an Asterisk
server and from an alternate point.
With Asterisk, I am hearing ringing and with the other origination point, I
am getting a message played on the far-end indicating to wait
Quoting Tilghman Lesher tles...@digium.com:
http://www.digchip.com/datasheets/parts/datasheet/222/82V2088-pdf.php
See pages 17-18 of the associated PDF. While this is not the T1 framer chip
used, the values are identical, which leads me to believe that these values
are actually industry
Hi Guys,
Running the following with a Sangoma A101D PRI card:
*Asterisk 1.4.21.2*
*LibPRI version: 1.4.10*
No inbound or outbound calls can be made. In fact Asterisk CLI doesn't show
any activity. Problem goes away on restart of the system or maybe asterisk.
I see post about upgrading Libpri to
Hello, all,
I have a Linksys 3102 from a VoIP provider. It use SRV record to
register to the provider's SIP server.
When I configure this line into my Asterisk, the register doesn't work
if I use their domain name.
So it like this:
If I use register = user:p...@proxy.provider.com
then I got:
Hi Guys,
I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the
current 1.4.10 version. I am running Asterisk 1.4.x (in fact it is a
PBXinaFLASH system).
How can I upgrade to the latest Libpri? Do I need to re-install Asterisk?
Won't that break the box?
Can I simply do this
This is another error msg from CLI:
[2010-05-17 11:48:50] WARNING[10957]: acl.c:400 ast_get_ip_or_srv:
Unable to lookup 'proxy.provider.com'
Jian Gao wrote:
Hello, all,
I have a Linksys 3102 from a VoIP provider. It use SRV record to
register to the provider's SIP server.
When I
hello,
still reports of non-updated dns caches in various sites of of the
world, so I redirected an older subdomain to the page:
http://ngs.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
Sorry for any inconvenience on the list,
Daniel
On 5/17/10 2:08 PM, Hristo Benev wrote:
On Mon, May 17, 2010 at 03:22:04PM -0400, bruce bruce wrote:
Hi Guys,
I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the
current 1.4.10 version. I am running Asterisk 1.4.x (in fact it is a
PBXinaFLASH system).
How can I upgrade to the latest Libpri? Do I need to
Thanks for the help Tzafrir.
I think for libpri you meant = 1.4.x rather than 1.4.4 as the latest
version available is 1.4.10.2or maybe 1.4.10 is greater than 1.4.4 ?!
Why haven't they changed the name to 1.5.0. I never get the nomenclature for
these things.
Thanks again,
Bruce
On Mon, May
I am announcing sound recognition project/library SoundPatty. It is
created to capture a recording in an audio stream. Use cases:
You can listen to live radio station and log how many advertisements
are played per day
You can know if leg B is an amazon.com bot :-)
You can match special operator
When I do show pri version I still see 1.4.10. Is that right or should I
see 1.4.10.2 since I upgraded it.
I did the install but shouldn't I get an Install sucesful message ?
*Just did a make clean make make install and output is:*
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm testing a telephone connected to FXS port of a Sangoma A200 card.
But I'm observing that callerid is not working. The configuration that
I'm using in chan_dahdi.conf is the following one:
-
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