On 04/13/2011 11:20 AM, Ishfaq Malik wrote:
On Wed, 2011-04-13 at 11:09 +0200, Jonas Kellens wrote:
On 04/13/2011 10:57 AM, Ishfaq Malik wrote:
On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote:
Hello,
I'm using SIP realtime with MySQL DB.
Is it possible to get the
Hi,
How to know the all SIP extensions status with AMI's ExtensionState ?
What is the value should I pass in
Context: ??
which will be define at context here ? shell I use sip.conf's context for
that extension or any other?
extension : ??
extension will be SIP/100 or just 100 ??
Please
Maybe I should have asked 'why do you want to put the status in to a
mySQL database'?
BTW - extensions.conf has mySQL functions built in - so no external
script is actually needed.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Wed, 2011-04-13 at 10:28 +0100, Andrew Thomas wrote:
BTW - extensions.conf has mySQL functions built in - so no external
script is actually needed.
Could you point me in the right direction for that?
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
On 04/13/2011 11:28 AM, Andrew Thomas wrote:
Maybe I should have asked 'why do you want to put the status in to a
mySQL database'?
BTW - extensions.conf has mySQL functions built in - so no external
script is actually needed.
Well, I read out this information in a website which serves as a
On Wed, 2011-04-13 at 11:24 +0200, Jonas Kellens wrote:
On 04/13/2011 11:20 AM, Ishfaq Malik wrote:
On Wed, 2011-04-13 at 11:09 +0200, Jonas Kellens wrote:
On 04/13/2011 10:57 AM, Ishfaq Malik wrote:
On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote:
Hello,
On Wed, 2011-04-13 at 10:32 +0100, Ishfaq Malik wrote:
On Wed, 2011-04-13 at 10:28 +0100, Andrew Thomas wrote:
BTW - extensions.conf has mySQL functions built in - so no external
script is actually needed.
Could you point me in the right direction for that?
Ignore that, I
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
And yes, I meant Asterisk has mySQL commands built in [that can be
accessed via. extensions.conf]. Sorry if I mislead.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Fair enough. Then if this is really what you want I guess an AGI is the
best way to go.
As for load - well, that depends on how many concurrent connections you
figure on having [and of course the platform it's all on].
-Original Message-
From:
On 04/13/2011 11:46 AM, Andrew Thomas wrote:
Fair enough. Then if this is really what you want I guess an AGI is the
best way to go.
As for load - well, that depends on how many concurrent connections you
figure on having [and of course the platform it's all on].
Platform : CentOS 5.5
Hi Russell,
Have you seen the 'Action URL' bit yet? Makes everything almost
key-system like ;)
BTW - one downfall of the Yealink is that it can't send different DND
commands to different accounts (it sends the one command to all
accounts). Not very useful if providers use different commands for
Hi
I believe I made one mistake in my example, I don't use a call to Queue
in my local channel without a partner channel (the customer). I'll
revisit this later today when I have some time, I'll be glad to help you
if I can recall the right solution :)
That would explain it. I wonder if we
Instead of picking from multiple scripts, send the action to the script
in a variable like the dial status
On 4/12/11 4:58 PM, Warren Selby wrote:
Sorry for the top post, on my phone...
It makes sense for someone who has written a custom visual voicemail
application to be able know when the
Hi,
I am working on integration of 2 systems: asterisk and messaging
platform. What I need is to access somehow information about current
calls. Should I do it over AMI ?
I need to be able to perform those 2 actions:
- How can I obtain msisdns of current calls ?
- How to hangup one of current
- Original Message -
I've just started deploying these (well the T28P model) after years
of
Snom issues and they look pretty good (although the documentation is
execrable; if you thought the Snom stuff was obtuse Yealink have got
them knocked into a cocked hat!).
Anyway, for
Asterisk as a phone system makes perfect sense in a condo. You can get
all the DID's you want and eliminate costs for the owners. You can offer
standard FXO for people who don't care and IP sets for people who want
to upgrade to feature sets.
Your door openner is a piece of cake.
1. Create an
I understand now the need for externnotify to run on vm check is to update
MWI. But I agree with M Hulber, an extra variable would be nice to tell the
script why externnotify was called...
On Wed, Apr 13, 2011 at 6:32 AM, M Hulber asterisk.ad...@hulber.com wrote:
Instead of picking from
Hi. I just want to make sure I understand this before doing something that
might break things spectacularly for our users and customers :)
We are using Asterisk 1.6.2.9 and my programming language of choice is Perl.
I want, when a call comes in on someone's DDI number (which the person who
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, April 13, 2011 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know extensions status ???
Am 13.04.2011 15:08, schrieb A J Stiles:
Hi. I just want to make sure I understand this before doing something that
might break things spectacularly for our users and customers :)
We are using Asterisk 1.6.2.9 and my programming language of choice is Perl.
I want, when a call comes in on
On Wed, 13 Apr 2011, A J Stiles wrote:
I want, when a call comes in on someone's DDI number (which the person
who dialled it can only possibly have obtained by dialling 1471 after we
called them), to be able to look up the caller's details from one of our
databases (where the number ought to
I have 2 separate Asterisk servers that are both exibiting this problem. 1
has a 4 port
FXO digium card, the other an 8 port.
For some reason when the machine reboots, the dahdi drivers are not properly
loaded. Then asterisk
ends up starting without dahdi support. I've tried everything that I
On Wed, 13 Apr 2011, Andrew Thomas wrote:
Fair enough. Then if this is really what you want I guess an AGI is the
best way to go.
As for load - well, that depends on how many concurrent connections you
figure on having [and of course the platform it's all on].
I do almost exactly the same
On 14/04/11 1:48 AM, Shawn L wrote:
I have 2 separate Asterisk servers that are both exibiting this
problem. 1 has a 4 port
FXO digium card, the other an 8 port.
For some reason when the machine reboots, the dahdi drivers are not
properly loaded. Then asterisk
To fix it, all i have to do is
Hi,
I have Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI
Express ) Card installed on the box. *Its not detected.* Details are as
below :-
[root@asterisk ~]# lspci
00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01)
00:01.0 PCI bridge: ATI Technologies Inc
Hi, I continue the discussion from
https://issues.asterisk.org/view.php?id=19103
If T.38 reinvite detection should still work, why it doesn't?
If I use faxdetect = t38 it does never detect the fax, even using alaw.
Cheers,
Darkbasic
--
On Wed, Apr 13, 2011 at 09:48:47AM -0400, Shawn L wrote:
I have 2 separate Asterisk servers that are both exibiting this
problem. 1 has a 4 port FXO digium card, the other an 8 port.
For some reason when the machine reboots, the dahdi drivers are not
properly loaded. Then asterisk ends up
Try dmesg command
root@:~# dmesg | grep -i Sangoma
[ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0 (c) 1994-2010
Sangoma Technologies Inc
[ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010
Sangoma Technologies Inc
[ 2303.494824] WANPIPE(tm)
make sure following init script is on at start up.
/etc/init.d/dahdi
Run lsmod command to make sure driver is loaded.
-S
Date: Wed, 13 Apr 2011 09:35:52 -0500
From: sruff...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problems With DAHDI on Ubuntu
Hi Guys!
I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it could
handle in production so following is my senario.
[sipp_client]---[Asterisk][sipp_server]
sipp_client
./sipp -sf uac_pcap.xml -d 10 -i 172.30.254.211 -s 2000 172.30.1.47 -l
On Wed, Apr 13, 2011 at 10:50 AM, satish patel satish...@hotmail.com wrote:
Hi Guys!
I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it
could handle in production so following is my senario.
[sipp_client]---[Asterisk][sipp_server]
Oops! Asterisk open 419 total thread and stop accepting connection. Can we
control number of thread to open or limit ?
root@:~# ps -C asterisk -L -o pid,tid,pcpu,state,nlwp,args | wc -l
419
Date: Wed, 13 Apr 2011 10:58:31 -0400
From: lath...@gmail.com
To:
On Wed, Apr 13, 2011 at 8:07 PM, satish patel satish...@hotmail.com wrote:
Try dmesg command
root@:~# dmesg | grep -i Sangoma
[ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0 (c) 1994-2010
Sangoma Technologies Inc
[ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0
Do you have the Sangoma wanpipe software installed?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 13, 2011, at 7:37 AM, satish patel wrote:
Try dmesg command
root@:~# dmesg | grep -i Sangoma
[ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0
On Wed, Apr 13, 2011 at 8:07 PM, satish patel satish...@hotmail.com wrote:
Try dmesg command
root@:~# dmesg | grep -i Sangoma
[ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0 (c) 1994-2010
Sangoma Technologies Inc
[ 2303.481115] WANPIPE(tm) Interface Support Module
Greetings Asterisk Users,
I'm happy to announce that Russell Bryant and Leif Madsen have volunteered to
host the next Asterisk Tech Tips webinar, next Thursday April 21 at noon
central time. Russell and Leif are project leaders and have collaborated on two
Asterisk books: Asterisk: The
Hello,
Some background before i ask the question.
I am attempting to implement a SIP trunk between an askerisk an a Mitel 5000
system. The mitel is giving me 404 errors when I send a call over to it even
though the call desination is valid. The mitel also reports an error saying
cp_dest_id
On 13/04/2011 10:14 PM, Niccolò Belli wrote:
Hi, I continue the discussion from
https://issues.asterisk.org/view.php?id=19103
If T.38 reinvite detection should still work, why it doesn't?
If I use faxdetect = t38 it does never detect the fax, even using alaw.
That is because the remote
Quoth Andrew Thomas:-
Have you seen the 'Action URL' bit yet? Makes everything almost
key-system like ;)
I saw it in the DSS key settings but havn't worked out anything useful
to do with it yet?
What are you using it for (and how?)?
--
Regards,
Russell
On Wednesday 13 April 2011 08:08:03 A J Stiles wrote:
Hi. I just want to make sure I understand this before doing something
that might break things spectacularly for our users and customers :)
We are using Asterisk 1.6.2.9 and my programming language of choice is
Perl.
I want, when a
On Wed, Apr 13, 2011 at 9:23 PM, Tim Nelson tnel...@rockbochs.com wrote:
On Wed, Apr 13, 2011 at 8:07 PM, satish patel satish...@hotmail.comwrote:
Try dmesg command
root@:~# dmesg | grep -i Sangoma
[ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0 (c) 1994-2010
Sangoma
On 4/12/11 1:21 AM, Don Kelly wrote:
Continuing top posting...
The same argument could be made for any commercial solution. Why use
Asterisk when we could throw $4,000 at our problem for a commercial
solution?
I'd like to have a solution that would have the features you suggest for
$400.
Rather than add extra overhead to your dialplan and the asterisk
server, why not make use of the AMI and have a background process
listening for the various events and updating your database accordingly
?
See
http://www.voip-info.org/wiki/view/asterisk+manager+events#ExtensionStatusEvent
and
Centos 5.6 came out. Any one tried to update to the 5.6 yet?
I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6?
--
*Jian*
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote:
Centos 5.6 came out. Any one tried to update to the 5.6 yet?
I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6?
I'm not sure about Asterisk in general, but if you use DAHDI, please be sure
to install version
Hello
Can some body let me know any softphone that is developed using java can
support at least sip protocol. Must be open source and ready to be used. I
am trying to accomplish is to integrate it with an applet. some thing like
click to call on web page.
Sorry if this is not correct place for
I just upgraded a non critical server.
Before upgrade: Centos 5.5(i386) + Dahdi 2.4.1 + Asterisk 1.8.3
First upgrade to Centos 5.6. After reboot, Asterisk is OK but Dahdi
failed!!!
Then I upgrade Dahdi to 2.4.1.2, also upgraded Asterisk to 1.8.3.2.
After reboot, Dahdi came back.
:)
*
Jian
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jian Gao
Sent: Wednesday, April 13, 2011 3:43 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???
I just upgraded a
On Wed, Apr 13, 2011 at 11:34 AM, Andreas Sikkema h...@ramdyne.nl wrote:
On 4/12/11 1:21 AM, Don Kelly wrote:
Continuing top posting...
The same argument could be made for any commercial solution. Why use
Asterisk when we could throw $4,000 at our problem for a commercial
solution?
I'd
I'd say boot with ubuntu livecd and run lspci command.
Make sure your card back side LED working.
--
Sent from my iPhone
On Apr 13, 2011, at 2:32 PM, Kaushal Shriyan
kaushalshri...@gmail.com wrote:
On Wed, Apr 13, 2011 at 9:23 PM, Tim Nelson tnel...@rockbochs.com
wrote:
On Wed, Apr
As far as you are not upgrading kernel you are good with Dahdi.
Kernel upgrade require to install dahadi
--
Sent from my iPhone
On Apr 13, 2011, at 4:04 PM, Shaun Ruffell sruff...@digium.com wrote:
On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote:
Centos 5.6 came out. Any one tried
Il 13/04/2011 19:54, Larry Moore ha scritto:
That is because the remote endpoint, eutelia, will need to detect the
Fax Tones and send the T.38 ReINVITE to you, they may not have T.38
enabled.
Uhm... it's very unlikely.
As a suggestion you could configure your incoming calls from eutelia to
Hi,
I have been trying to find out what module is causing asterisk to open port
5000
I have already disabled some ( sccp, mgcp, iax and other modules ) since I
only want sip port opened
/etc/asterisk# netstat -aln --programs | grep asterisk
tcp0 X.X.X.X:5060 0.0.0.0:*
http://www.google.com/search?q=port+5000+asterisk
answer is in the first hit :)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Thomas
Sent: Wednesday, April 13, 2011 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial
Is there an way (asteisk command / AMI / Agi ) to process incoming SIP
messages like ( 100 trying , 183 session progress , 200 Ack) ,
I am intersted to findout delay between 183 and 200 message
Regards
Nasir Iqbal
--
_
--
Though you can modify SIP headers, there is no straight forward way of
processing SIP messages as Asterisk is made to abstract away that protocol
layer. However it generally is possible to track the Ringing and Answer
events in AMI which in turn the reflect the 180/183 and 200 SIP messages.
How to use telnet ? I never work on that . please guide me...
On Wed, Apr 13, 2011 at 6:48 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra
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