Hello,
I notice that the following proces is running :
astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527
What is this ??
Kind regards,
Jonas.
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On 24 November 2010 01:20, Lyle Giese l...@lcrcomputer.net wrote:
Post the revelent portions of your extension.conf. Maybe you have a logic
error somewhere.
Thanks Lyle.
My extensions.conf is fairly simple in this regard; I use macro-stdexten:
[macro-stdexten];
exten =
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
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We use it to determine who is the caller.
Regards,
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Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
- Original Message -
Hello,
I notice that the following proces is running :
astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527
What is this ??
Kind regards,
Jonas.
You are running Asterisk with priority set. Check /etc/asterisk/asterisk.conf
for the
No you can't
On Wed, Nov 24, 2010 at 2:34 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
On Wed, Nov 24, 2010 at 1:20 AM, Hans Witvliet h...@a-domani.nl wrote:
Hi all,
Perhaps someone has dealt with it before.
I want to activate a bunch of my own scripts after someone has registred
om my asterisk, or when his cient has de-registerded.
have been skimming through AGI and AMI,
On 11/24/2010 10:28 AM, --[ UxBoD ]-- wrote:
Hello,
I notice that the following proces is running :
astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527
What is this ??
Kind
I am confused. In Asterisk 1.2 and 1.4, in the code there is an error:
Setting a group requires an argument (group name)
But the syntax is shown as: Syntax: GROUP([category])
The [category] square brackets indicate to me an optional parameter,
which contradicts the error.
Verison 1.6 is
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
in extensions.conf:
exten = 0532xx,1,Answer
exten = 0532xx,2,MusicOnHold(Sound_1)
exten = 0532xx,3,Dial(SIP/ACCOUNT001,180,t)
exten =
On 24 November 2010 10:12, Steve Davies davies...@gmail.com wrote:
I am confused. In Asterisk 1.2 and 1.4, in the code there is an error:
Setting a group requires an argument (group name)
But the syntax is shown as: Syntax: GROUP([category])
The [category] square brackets indicate to me an
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
in extensions.conf:
exten = 0532xx,1,Answer
exten =
Hi all,
I want to upgared from asterisk-1.6.2.6 version to asterisk-1.8.0 version.
When i execute make command for compilation i have seen below errors.
In file included from
/usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/cdr.h:31
Hello all,
I am using Linksys SPA942 in my current installation activity. I see a
peculiar behavior: A call is made and the SPA942 uses its speaker. When the
far end of a call hangs up , the SPA942 stays off hook, and after a time
plays a fast busy. The user then has to press the line presence
Hello,
it's my first post on this list, I hope not to bore youwith my novice
questions..
We're using a TDM400 with 3 fxo modules connected to pstn.
Call goes inbound/outbound correctly, after playing a bit on some
dahdi-channels.conf/chan_dahdi.conf options.
The big problem is that after 5
2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com:
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
in
On Wed, Nov 24, 2010 at 5:16 AM, Cassius Smith cass...@cassius.org wrote:
Hello all,
I am using Linksys SPA942 in my current installation activity. I see a
peculiar behavior: A call is made and the SPA942 uses its speaker. When the
far end of a call hangs up , the SPA942 stays off hook, and
On Wed, Nov 24, 2010 at 5:35 AM, Olivier CALVANO o.calv...@gmail.com wrote:
2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com:
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
Hi,
Does anyone know if CID is already working with Digium TDM800P card
using DTMF signalling?
(I'm brazillian)
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New to Asterisk? Join us for a live
My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
is :
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x861f6d8', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x85a6420', 9
On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
should be easy enough to test.
Here is an example of what I see on the manager interface during a
register/unregister:
Event: PeerStatus
Privilege:
It is the phone itself: go to Regional tab and scroll down to Reorder Delay
and make it 255. That tells it not to play re-order tone and just hangup.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith
Sent: Wednesday,
Hi to all.
I am conducting several tests with the Asterisk manager and I noticed
something that I believe to be a problem.
When I generate a call with the Action Originate with the Async option true,
the event OriginateResponse returns normally. But when I generate a call in
the same way,
On 10-11-24 06:09 AM, RAJNIKANT VANZA wrote:
make[1]: *** [cdr_webservice.o] Error 1
make: *** [cdr] Error 2
What is cdr_webservice.o ?
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
2010/11/20 Olivier oza_4...@yahoo.fr
Depending on what telco Charlie is connected to would change the CallerId
presented to Charlie from being Alice's or Bob's Cid.
When a call is forwarded, Charlie's telco receives different ANI and CID :
some (seems to) favor ANI and some CID.
An
Hello list,
I'm experiencing a lot of server freezes lately. The server just... freezes.
I notice in the log files (/var/log/asterisk/messages
/var/log/messages) that logging stops at the time the server hangs.
Logging continues when the server has been restarted (which is the only
Yes, the thanksgiving holiday is here (in the USA)! But also, the fear
of running out of IP addresses next year has raised its ugly head and
since we don't do Thanksgiving in Europe, we have some serious talking
to do about this problem.
This Friday at 12 Noon EST, Olle Johansson will be joining
Hi Everyone,
I am wondering why documentation of some of the vital parts of Asterisk is
hosted on voipinfo.org (unreliable is some parts) and not on asterisk.org?
For example the list of AMI events are not well documented and one has to
guess which version supports which event. The documentation
On Wed, Nov 24, 2010 at 11:43 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Hello list,
I'm experiencing a lot of server freezes lately. The server just... freezes.
I notice in the log files (/var/log/asterisk/messages /var/log/messages)
that logging stops at the time the server hangs.
On 11/24/2010 10:43 AM, Jonas Kellens wrote:
Hello list,
I'm experiencing a lot of server freezes lately. The server just... freezes.
I notice in the log files (/var/log/asterisk/messages
/var/log/messages) that logging stops at the time the server hangs.
Logging continues when the
On 11/24/2010 11:02 AM, Shaun Ruffell wrote:
On 11/24/2010 10:43 AM, Jonas Kellens wrote:
The only thing I have is a high level of mentionning of kernel: dahdi:
Detected time shift. in /var/log/messages.
What is causing this kernel message ? Could this be the cause of the
server freeze ?
I know there was a patch for dahdi to fix server lockups on time shift. (not
sure what version, but if you changed the time, the server would just go
crash.)
Do you have the latest version ?
Check your ntpd settings to make sure your time isn't bouncing all over the
place.
Hi,
Starting in Asterisk 1.8.0, Asterisk supports connected line updates.
This is fantastic for SIP. How can I prevent them from being sent to a
PRI channel?
I'm having problems when a call is answered by an internal SIP
extension, then transferred (blind or attended) to another internal SIP
On Sun, Nov 21, 2010 at 11:13:00PM +, Jonathan Hunter wrote:
Hi,
I've been experiencing trouble with my DAHDI channels for some time and have
finally decided to try and resolve the issue.
Essentially, the problem I am having is that when a call comes in, and my
DAHDI phones therefore
On 23 Nov 2010 at 16:54, Joseph (Joseph syscon...@gmail.com) commented about
Re: [asterisk-users] Someone has hacked into our :
On 11/23/10 14:18, Gary Kuznitz wrote:
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk-
On Fri, 19 Nov 2010 10:15:40 -0500, jon pounder j...@inline.net
wrote:
What is nice is when the $50 hardware and the $1000 hardware run exactly
the same software so other than the drivers for the hardware itself,
everything else behaves the same way and its easy to move around
configurations to
Thank you for the reply.
On 23 Nov 2010 at 18:51, John (John Novack jnov...@stromberg-carlson.org)
commented about Re: [asterisk-users] Someone has hacked into our :
Gary Kuznitz wrote:
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman
On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
should be easy enough to test.
Here is an example of what I see on the manager interface during a
On 11/24/10 10:39, Gary Kuznitz wrote:
Look for allowguest default is yes
I change it to allowguest=no
In addition you might want to restrict some countries in your dial-plan,
here is my list:
This would be great. Can I put this anyplace in extensions.conf?
Or does it need to go after
On 10/14/10 15:38, Bryant Zimmerman wrote:
For which device models?
From: Mark Murawski markm-li...@intellasoft.net
Sent: Thursday, October 14, 2010 3:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Audiocodes firmware
Does anyone
On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet h...@a-domani.nl wrote:
On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
should be easy enough to
Am 24.11.2010 13:48, schrieb Bayardo Sanchez:
My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
is :
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x861f6d8', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780
Othe problem is small time my hdd is full of recording
--
Sent from my BlackBerry®
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US Numbers: 561-886-0664
Nicaragua Mobile: +505.8488.6876
-Original Message-
From: Stefan Schmidt s...@sil.at
Sender:
On Wed, 2010-11-24 at 15:47 -0600, Sherwood McGowan wrote:
On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet h...@a-domani.nl wrote:
On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
a PeerStatus event. I don't know if
On Wed, Nov 24, 2010 at 4:24 PM, Hans Witvliet h...@a-domani.nl wrote:
On Wed, 2010-11-24 at 15:47 -0600, Sherwood McGowan wrote:
On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet h...@a-domani.nl wrote:
On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
On Asterisk 1.8 when a SIP peer
Tzafrir,
On 24 November 2010 18:12, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Can you replicate those phantom answers without calling all channels?
Try:
originate DAHDI/7 application Echo
Does that line answer without you picking up the phone? Or does it
require a combination of
On Wed, Nov 24, 2010 at 11:30 AM, Tilghman Lesher tles...@digium.comwrote:
On Wednesday 24 November 2010 11:07:40 Bruce B wrote:
This is not to bash the Asterisk project or Digium. Don't respond if you
have a difference of opinion as I am not looking for personal opinions
but rather JUST
Hello,
We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.
The Telco they were buying the trunks to discovered this configuration and
restricted
On Wed, Nov 24, 2010 at 7:20 PM, Warren Selby wcse...@selbytech.com wrote:
On Wed, Nov 24, 2010 at 11:30 AM, Tilghman Lesher tles...@digium.com
wrote:
On Wednesday 24 November 2010 11:07:40 Bruce B wrote:
This is not to bash the Asterisk project or Digium. Don't respond if you
have a
I am not sure where you are and what legal conventions are involved.
Are you saying the Telco (and legal restrictions) say you cant send calls
to the internet via the AS5300 but you can if Asterisk does it directly?
What is the logic in that?
Or are they saying your Telco to Asterisk
On 10-11-24 08:34 PM, Sherwood McGowan wrote:
True, but then some of us registered on that site and still don't have
the ability to edit...I thought it was a community effort? Maybe I was
wrong
Once registered you will be able to post comments, not edit. If you
would like to become part of
On Wed, Nov 24, 2010 at 8:06 PM, Paul Belanger pabelan...@digium.com wrote:
On 10-11-24 08:34 PM, Sherwood McGowan wrote:
True, but then some of us registered on that site and still don't have
the ability to edit...I thought it was a community effort? Maybe I was
wrong
Once registered you
Thanks Cary,
What happens is, the Telco won't allow the small company to resell the ISDN
connections, meaning, they bought the trunks and DIDs, then sold dialing
plans to route incoming calls through the PRIs out the Internet. This is not
the issue though. We definitely have to migrate to an SS7
Jonathan Hunter wrote:
On 24 November 2010 01:20, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Post the revelent portions of your extension.conf. Maybe you have
a logic error somewhere.
Thanks Lyle.
My extensions.conf is fairly simple in this regard; I use
I have been pounded with new, mostly text spam in the last few weeks.
Tonight I realized that the address that is being spammed is a personal one
I use for this list.
Has anyone else noticed new spam in the last 2-3 weeks?
Cary Fitch
--
Thanks Cary,
The first topology we are working on should be the best way then.
Asterisk will answer SS7 calls, route them to the ISDN channels to be
terminated by the AS5300 as they were doing before. I think TDM-2-TDM
shouldn't be that much of a problem eh? No further equipment needed?
*José
Cary Fitch wrote:
Has anyone else noticed new spam in the last 2-3 weeks?
No,
But I run ASSP in front of my MTA.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
Same here. But, can the genie ever be put back in the bottle?
Cary Fitch wrote:
Has anyone else noticed new spam in the last 2-3 weeks?
No,
But I run ASSP in front of my MTA.
Doug
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-- Bandwidth and Colocation
Thank you Horacio and Cary.
We will try receiving SS7, routing via SIP, answering on the AS5300, then
looping back to itself (out PRI, in PRI ports) in order to invoke the modem
termination. This way we may be able to spare the TDM cards in Asterisk and
reuse the E1 ports installed in the
Hi Paul,
Thanks for reply.
I have some mistake send compilation logs. i have written cdr_webservice.c
module and its work on asterisk-1.6.2.6 version on production server. but i
want to upgrade asterisk version.
# make
[CC] cdr_webservice.c - cdr_webservice.o
In file included from
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