Hello,
Is there easiest way to count active channels per peer (peername) using
AMI or CLI? The hard way would be : check all active channel and get their
information : sip show channel 24609-fk-016ebea9-6e1e59...@10.10.10.1 and
we get Peername .
Thanks
--
Best Regards,
Giedrius
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On Mon, Feb 01, 2010 at 07:42:51AM +, frangky robert wrote:
I do some test:
1.unplug usb connector from server to astricon
2.unplug power to astricon
3.plug-in the power to astricon
4.plug-in the usb connector
Here is the log from /var/log/messages after doing the 1st step.
On 02/01/2010 08:38 AM, randall wrote:
hi all,
just had a terrible and sleepless weekend at the office trying to get
asterisk going, its just tough love ;)
have tried several asterisk versions but i currently have the following
setup on debian lenny that kind of works.
asterisk-1.6.2.0
Dear List,
i have been thinking of building a calling cards solution based on Asterisk and
a2billing..
i have a few questions regarding this solution and was hoping you may have the
answers and could be generous enough to offer them.
the servers i'm thinking of are with the following Specs:
It entirely depends on the technology used to interface to the PSTN.
You have not specified what technology/hardware you are using to connect
to the PSTN.
For instance if you are using POTS(plain old telephone service - analog
copper fed lines), you do not get answer supervision back from the
On Mon, Feb 01, 2010 at 08:38:36AM +0100, randall wrote:
hi all,
just had a terrible and sleepless weekend at the office trying to get
asterisk going, its just tough love ;)
have tried several asterisk versions but i currently have the following
setup on debian lenny that kind of works.
Hi- can anyone help with this. I'm really stuck as apparently it
should work. Is it a problem with the ITSP, with using the same trunk
for both legs of the call etc?
John
On 30 January 2010 08:57, John Taylor j...@vetsurgeon.org.uk wrote:
Hi
If I have an incoming call coming down a SIP trunk
On 26 January 2010 04:21, Joel Lansden j...@digitalparadise.net wrote:
Greetings all.
First off, thank you for your time on this. I have spent literally 12 hours
searching every forum and article I can find, and I’m going cross-eyed, so I
need to bother everyone with this.
I am running *
hi,
you can lookup the causes in the sources
check you dahdi-configuration (especially the groups...) is there
everything ok? what does dahdi_tools or the other cli-commands say, that
give you
information about the available channels?
yves
/* Causes for disconnection (from Q.931) */
Any idea what can cause this?
asterisk*CLI core show channels
Channel Location State Application(Data)
Logger/rotates...@default:1 Down(None)
1 active channel
0 active calls
20229 calls processed
asterisk*CLI
Hi all,
Do anyone has a detailed procedure for NV_application install? I have search
as I was told, but I did no find any thing accurate.
Thanks
ML
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Hi all,
I know this could sound like a ghost story but...
I have prepared an Asterisk 1.4.26.2 PBX on a Debian Lenny, the same I
always prepare, same hw, same sw. I connected it to 2 iax phones using a
hub: it works!
I take everything it to our customer place, same PBX, same phones, same
hub,
1.4.x doesn't natively support logging the queue to to mysql, and can't find
any way without batching it from the /var/log/asterisk/queue
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
Sent:
Hi,
I'm using the default Asterisk function Monitor to record calls, but i have
some issue's with this, the problem is when a call is finished, it never mix
in out together, bellow you can see my call configuration:
exten = _8.,1,Monitor(wav,${EXTEN},m)
exten =
Yes, over asterisk 1.4.x you need put an cron or deamon to load queue
log into database. I suggest check on the list for more details.
Regards,
On Mon, Feb 1, 2010 at 8:51 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
1.4.x doesn't natively support logging the
There's nothing wrong with this per se; it just needs to be in a context;
try it this way;
- exten = 8284,1,Goto(domath,s,1)
[domath]
Exten = s,1,play(to-call-num-press)
- exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
- exten = 4,n,WaitExten(3)
- exten = 4,n,Goto(domath,s,1)
- exten =
Uros Djokic wrote:
It entirely depends on the technology used to interface to the PSTN.
You have not specified what technology/hardware you are using to connect
to the PSTN.
For instance if you are using POTS(plain old telephone service - analog
copper fed lines), you
This is my feeble attempt at explaining this;
DTMF is processed by two dial-plan commands that I'm familiar with (there
are more but I'm trying to speak from my knowledge)
Waitexten reads 1 or more DTMF digits and goes to an extension in the
dialplan.
Example
- exten = s,1,playback(welcome)
-
Thank you.
I was also thinking of using the READ application to store dtmp variabes.
Then total them up at the end.
More to follow.
P
On Mon, Feb 1, 2010 at 9:20 AM, Danny Nicholas da...@debsinc.com wrote:
There's nothing wrong with this per se; it just needs to be in a context;
try it this
Here's how I would do this based on the post below (it's below in outlook
express)
Exten = 866,1,Goto(tommath,s,1)
[tommath]
- exten = s,1,Read(NUMBER,instruct,2,skip,5)
- exten = s,n,Gotoif($[${NUMBER} = 1]?one)
- exten = s,n,Gotoif($[${NUMBER} = 2]?two)
- exten = s,n,Gotoif($[${NUMBER} =
On 02/01/10 09:54, Mariano Lecuona wrote:
Hi all,
Do anyone has a detailed procedure for NV_application install? I have search
as I was told, but I did no find any thing accurate.
Thanks
ML
I was running it with Asterisk-1.2 (nice application) one would think that it
should be incorporate
Joseph wrote:
I was running it with Asterisk-1.2 (nice application) one would think that it
should be incorporate into asterisk but they are not doing it, nor the
developer
is interesting in upgrading the code ever-time there is a new version of
asterisk and something is broken.
I was
On 02/01/10 12:01, Kevin P. Fleming wrote:
Joseph wrote:
I was running it with Asterisk-1.2 (nice application) one would think that
it should be incorporate into asterisk but they are not doing it, nor the
developer
is interesting in upgrading the code ever-time there is a new version of
YMMV, but NVFaxdetect can be activated in 1.4.x;
Try this link
http://blogtech.oc9.com/index.php?option=com_contentview=articleid=77%3A20
071121astItemid=8
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On Mon, Feb 1, 2010 at 8:55 AM, Peter den Hartog
peterdenhar...@gmail.com wrote:
I'm using the default Asterisk function Monitor to record calls, but i have
some issue's with this, the problem is when a call is finished, it never mix
in out together, bellow you can see my call configuration:
This is how I understand it. The other end is trying to set up comfort
noise and asterisk is letting you know that it's trying to do so and
maybe you can turn this off on the other end. I have a particular voip
provider where I get this message. I think if you get it turned off
there's a
Hello list !
I'm having one way audio on incoming and outgoing calls. Outgoing audio
works fine, incoming audio is not working.
My setup is the following :
incoming calls :
PSTN -- FXOport -- HT503 -- WANport -- Asterisk -- WANport -- HT503 (the
same) -- FXSport -- DECTphone
outgoing calls :
On 31/01/2010 04:35, Tzafrir Cohen wrote:
Yes, please see https://issues.asterisk.org/view.php?id=16493
Basically the driver needs minimal fixing. Probably just to add the PCI
ID to the list.
Hi,
Thanks
Gonna have a look at it.
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Please descard me from the asterisk users list...thanks
(Abu Nasar Mahmud)
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Is there a way to make a virtual extension busy programmatically?
I want to be able to turn lights on and off on a Polycom phone from a script.
-Matt
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Is there a way to make a virtual extension busy programmatically?
I want to be able to turn lights on and off on a Polycom phone from a script.
That's what the Custom device type is for.
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Richard Kenner wrote:
Is there a way to make a virtual extension busy programmatically?
I want to be able to turn lights on and off on a Polycom phone from a script.
That's what the Custom device type is for.
please elaborate I would like to know too
--
In your dialplan, you should put in...sth like
exten = 1001,hint,Custom:virtext1001
In your script, you should put in...sth like
Set(DEVSTATE(Custom:virtext001=INUSE);
Set(DEVSTATE(Custom:virtext001=NOTINUSE);
In the phone directory.xml, define an entry with ct=1001 and turn bw on.
Reboot phone
If you read your message all the way to the end, and every posting, you
will discover exactly how to do that on your own.
asterisk-users mailing list To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
That's what the Custom device type is for.
please elaborate I would like to know too
See http://www.voip-info.org/wiki/view/Asterisk+func+device_State
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How small can an Asterisk system be, in terms of disk space utilized?
I am looking for just asterisk, with mysql, postgresql, or sqlite,
with PHP and Python.
After finishing the build and removing the tools, how small can the
whole system be?
100Mb, 200Mb?
Can packages be used to build the
I think Astlinux comes in under 100MB.
Ben M. Schorr
Chief Executive Officer
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Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
Twitter: http://www.twitter.com/bschorr
Facebook: http://www.facebook.com/rolandschorr
-Original
On Tue, Feb 2, 2010 at 6:41 AM, Frank Church voi...@googlemail.com wrote:
How small can an Asterisk system be, in terms of disk space utilized?
I am looking for just asterisk, with mysql, postgresql, or sqlite,
with PHP and Python.
AskoziaPBX is 30MB installed. It has Asterisk 1.6.1, nearly
AstLinux is well under that. You could build a custom image that
contains only what you want and have it under 30M. We have support for
sqlite3, but not mysql or postgresql. You would have to build your own
package to include python. Our build environment is based on buildroot,
but has
hi,all
all realtime queue work fine except one thing:
in the queue_table ,when i change strategy from ringall to linear need
asterisk to restart!
[Feb 2 15:41:51] WARNING[4106]: app_queue.c:1532 queue_set_param:
Changing to the linear strategy currently requires asterisk to be
restarted.
i
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