what mean privilege: call, all for event? this is call or all?
why have two privilege
And how i can prevent manager to send me NewExt event?___
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asterisk-users mailing list
To
Hi to all,
The problem was finally solved by installing Asterisk-1.4.18. Versions 1.4.20
and current (1.4.22) produce the issue. Version 1.4.18 does not. May this be a
bug?
Aldo Sudak___
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Title: Re: [asterisk-users] Is SIPPEER curcalls working for you ?
Did You triedhttp://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers?
Hi,
In this threadhttp://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html, I wondered whether SIPPEER curcalls was working.
I
Good Day,
I have been tasked with fixing the time on our asterisk server. I am having
a hard time finding documentation to tell my what asterisk uses to get its
time information to push to phones (or a better question, where does the
SPA-962 get its time information)?
Basically, I can go under
http://bugs.digium.com/view.php?id=13645
Hi list,
I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2.
To cut a long story short, IAX2 is not tx-ing hangup...
Scenario is composed of two asterisk systems A and B.
A receives calls from IAX users X, Y, Z, etc, does some
validation
I've never used the Sipura phones but they probably sync with an NTP server.
My guess is that the NTP server is on the asterisk box (you can probably verify
this by checking the config of the phones and finding the option for NTP
server). It is possible that the NTP service isn't running on the
This Friday's edition of the weekly VoIP Users Conference call is all
about wideband audio (aka HD Voice) and conferencing. The guest for
this call is David Frankel, CEO of ZipDX a commercial service that
specializes in wideband conferencing. We expect an interesting call
touching on many aspects
2008/11/4 Igor Zamocky [EMAIL PROTECTED]
Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers
?
I didn't.
Now I did and it's working the way I wanted.
Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and
SIPPEER but limitonpeers is much more
On Tue, Nov 04, 2008 at 09:34:18AM -0600, Michael Graves wrote:
This Friday's edition of the weekly VoIP Users Conference call is all
about wideband audio (aka HD Voice) and conferencing. The guest for
this call is David Frankel, CEO of ZipDX a commercial service that
specializes in wideband
On Tue, 4 Nov 2008 17:48:58 +0200, Tzafrir Cohen wrote:
Is there any decent free soft phone that is not capable of Speex/wb?
A short check on my system: supporting:
ekiga 2.0.12-1+nmu1
linphone 2.1.1-1+b1
twinkle1:1.2-3
Not supporting:
iaxcomm2.0.2-3
But then
On 4 Nov 2008, at 14:17, Steve Anness wrote:
Duplicated stuff
Just the one post is usual sufficient.
On topic - I would check what the NTP server is set to on the SPA.
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Hi,
It does not appears to be a session-timers issue. There are no SIP
exchange except for BYE message initited after 20 minutes.
I increased the session timer to 3600 seconds and also tried your
suggestion without any luck. Any other inputs?
Thanks
Jim
On Mon, Nov 3, 2008 at 10:47 PM, John
On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves [EMAIL PROTECTED] wrote:
In any case, the wideband bridge for this weeks VUC call supports only
G.722.
But we do plan to make a recording of both conference version available, AFAIK?
r
___
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randulo wrote:
On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves [EMAIL PROTECTED] wrote:
In any case, the wideband bridge for this weeks VUC call supports only
G.722.
But we do plan to make a recording of both conference version available,
AFAIK?
r
But will it be a high-def
On Tue, 4 Nov 2008 17:10:42 +0100, randulo wrote:
On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves [EMAIL PROTECTED] wrote:
In any case, the wideband bridge for this weeks VUC call supports only
G.722.
But we do plan to make a recording of both conference version available, AFAIK?
r
Is there a way to override sip peers defined in users.conf with respect to
their context and hints?
Every extension I have defined in users.conf always gets an @default for the
hint priority. Below are asterisk outputs and users.conf entries. In peer
1203 I've set a subscribecontext, which
On Tue, 4 Nov 2008, Michael Graves wrote:
Also, a narrowband recording. It'll be interesting to hear if the
people connected via G.722 sound appreciably better in the narrowband
recording.
Will you be able to calibrate volume and frequency response (within
G.711 limits) between the
Dear List
I'm asking if there is a small hardware already implemented with
software that just do traffic Shaping QoS?
I have found *D-Link DI-102 VoIP QOS Adapter Packet Prioritizer
*But it does not look available in Italy !
Regards Andrea
**
Kristian Kielhofner ha scritto:
On Jan 9, 2008 11:44
On Tue, 4 Nov 2008 08:42:29 -0800 (PST), Steve Edwards wrote:
On Tue, 4 Nov 2008, Michael Graves wrote:
Also, a narrowband recording. It'll be interesting to hear if the
people connected via G.722 sound appreciably better in the narrowband
recording.
Will you be able to calibrate volume and
Steve Anness wrote:
Good Day,
I have been tasked with fixing the time on our asterisk server. I am
having a hard time finding documentation to tell my what asterisk uses
to get its time information to push to phones (or a better question,
where does the SPA-962 get its time information)?
On Tue, Nov 4, 2008 at 10:34 AM, Michael Graves [EMAIL PROTECTED] wrote:
This Friday's edition of the weekly VoIP Users Conference call is all
about wideband audio (aka HD Voice) and conferencing. The guest for
this call is David Frankel, CEO of ZipDX a commercial service that
specializes in
Kristian, heh...
In a word:
They've got a great product. Whether you actually want to go read ANY
company's BS ^H^H web site is another story! I've rarelty if ever seen
a web site with anything other than gobbledy gook text, but so few
people read the text in a web site, it matters little.
As
Just saw from build 2036:
Starting mini_httpd...
WARNING WARNING WARNING
YOU STILL HAVE NOT CHANGED YOUR HTTPS ADMIN PASSWORD
ANYONE THAT KNOWS YOU ARE USING ASTLINUX CAN DESTROY YOUR
SYSTEM. PLEASE CHANGE THIS OR DISABLE THE HTTPS ADMIN
INTERFACE IMMEDIATELY!
Example:
htpasswd
Philip Prindeville wrote:
Just saw from build 2036:
Now, to get the following packages to build:
misdn
asterisk-chanmisdn
nistnet
rhino
strace
rp-pppoe
Whoops. I'm sure Philip thought he was sending this to a different
mailing list.
Hi,
In this thread
http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html ,
I wondered whether SIPPEER curcalls was working.
I could test this anew today. Here are my findings :
Alice, Bob and Carol ar all using SIP Phones.
Whenever Alice is calling Bob,
- if Carol is
Darrick Hartman wrote:
Philip Prindeville wrote:
Just saw from build 2036:
Now, to get the following packages to build:
misdn
asterisk-chanmisdn
nistnet
rhino
strace
rp-pppoe
Whoops. I'm sure Philip thought he was sending this to a different
mailing list.
Hi,
We have been selling * systems for a while and always have used other
companies for origination and termination and let the client pay directly.
Since we do not have enough traffic to justify building our own
infrastructure we would like to start reselling someone else's service.
Any ideas? It
Hi. I am using gentoo kernel 2.6.27-r2 and dahdi trunk svn 5211 and
it will not compile with this kernel whereas it does compile with
2.6.25. Here is the relevant portion of the build:
VERIFY /usr/src/dahdi-trunk/drivers/dahdi/xpp/init_card_1_30
VERIFY
On Tuesday 04 November 2008 12:37:03 Leah Newmark wrote:
Thanks!
Apparently, I forgot to mention our version running is 1.2.13
Is the coding similar enough to give it a shot?
No, the code is likely to be completely different. In fact, the trouble
you're having may just be the reason why it
OK, thanks.
on Tuesday 11/04/2008 Tzafrir Cohen([EMAIL PROTECTED]) wrote
On Tue, Nov 04, 2008 at 03:17:30PM -0500, John covici wrote:
Hi. I am using gentoo kernel 2.6.27-r2 and dahdi trunk svn 5211 and
it will not compile with this kernel whereas it does compile with
2.6.25. Here is
Other things to check:
- ensure you don't have media timeouts set (see rtptimeout in
sip.conf)
- do you have any absolute timers set in your dialplan? (any $
{TIMEOUT(absolute)} values set?
- does your upstream carrier have any sort of timer limit? (try
using just a vanilla softphone
On Nov 4, 2008, at 6:08 AM, Steve Anness wrote:
Good Day,
I have been tasked with fixing the time on our asterisk server. I
am having a hard time finding documentation to tell my what asterisk
uses to get its time information to push to phones (or a better
question, where does the
On Tue, Nov 04, 2008 at 03:17:30PM -0500, John covici wrote:
Hi. I am using gentoo kernel 2.6.27-r2 and dahdi trunk svn 5211 and
it will not compile with this kernel whereas it does compile with
2.6.25. Here is the relevant portion of the build:
VERIFY
On 5/11/2008 5:57 a.m., andrea wrote:
Dear List
I'm asking if there is a small hardware already implemented with
software that just do traffic Shaping QoS?
I have found *D-Link DI-102 VoIP QOS Adapter Packet Prioritizer
*But it does not look available in Italy !
Regards Andrea
The Linksys
Hi list,
I'm wondering if there's a way for multiple users to share the same voicemail
box and have their BLF flashing when voicemail comes, i.e. in a home phone
system where there's a general vm for everyone.
I'm using couple Grandstream GXP2020.
Any suggestions?
Kelvin Chan |
Am Montag, den 03.11.2008, 13:20 + schrieb Steve Davies:
I found this only last week... The problem is not the short timer, it
is the dialtone audio definition (top of the same page IIRC). If you
look at the tone definition for Dialtone it is only a few seconds
long. When it runs out, the
On Tue, Nov 4, 2008 at 3:12 PM, Igor Zamocky [EMAIL PROTECTED] wrote:
http://bugs.digium.com/view.php?id=13645
Thanks Igor, we'll keep an eye on it.
--
exvito
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Simply put the shared mailbox on all the phones definition like:
mailbox=100,101
That way the phone will flash if there is mail on any of the boxes.
On Tue, 2008-11-04 at 13:31 -0800, Kelvin Chan wrote:
Hi list,
I'm wondering if there's a way for multiple users to share the
Did you know that any commandyou type in asterisk cli starting with
exclamation point (!) is execute in the shell by asterisk ??
Example :
running
!ls
will run 'ls' in your current directory
So, be aware because your user can do whatever we want then.
Dima wrote:
On Sat, Nov 01, 2008 at
Thanks!
Apparently, I forgot to mention our version running is 1.2.13
Is the coding similar enough to give it a shot?
(Perhaps it's time to upgrade...)
I really appreciate the assistance.
LN
[EMAIL PROTECTED] wrote:
Message: 12 Date: Mon, 3 Nov 2008 15:11:10 -0600 From: Tilghman Lesher
On Tue, 4 Nov 2008, Dima wrote:
The person I'm giving the access to is an admin of that asterisk. It's
up to him to do evil stuff with asterisk itself. as long as he can't get
a shell and do rm -rf / I'm safe.
Hmm, I wonder if you could run asterisk in a jail? Anyone done that on
FreeBSD
Good Day,
I have been tasked with fixing the time on our asterisk server. I am having
a hard time finding documentation to tell my what asterisk uses to get its
time information to push to phones (or a better question, where does the
SPA-962 get its time information)?
Basically, I can go under
On Sat, Nov 01, 2008 at 12:38:52AM +0100, Dima wrote:
Setting the user's shell to /usr/sbin/rasterisk works. On login user
gets into asterisk CLI if asterisk is running (user just has to have
write permission to /var/lib/asterisk.*).
How does that user login?
client$ ssh [EMAIL
It was a lack of free space in disk, because a big load of recorded calls
and logs.
Daniel
On Thu, Oct 23, 2008 at 12:40 PM, Daniel - Asterisk [EMAIL PROTECTED]wrote:
I'm restarting my system without solution and I've extended my call limit
to 10 calls (asterisk.conf) to avoid call
On Tuesday 04 November 2008 15:52:10 Ruddy Gbaguidi wrote:
Did you know that any commandyou type in asterisk cli starting with
exclamation point (!) is execute in the shell by asterisk ??
Example :
running
!ls
will run 'ls' in your current directory
So, be aware because your user can do
On Tuesday 04 November 2008 16:02:40 Jeff LaCoursiere wrote:
On Tue, 4 Nov 2008, Dima wrote:
The person I'm giving the access to is an admin of that asterisk. It's
up to him to do evil stuff with asterisk itself. as long as he can't get
a shell and do rm -rf / I'm safe.
Hmm, I wonder if
On Tuesday 04 November 2008 10:30:10 Jeremy Mann wrote:
Is there a way to override sip peers defined in users.conf with respect to
their context and hints?
No, there is not. Users.conf is meant to be a very simple interface for
adding users, designed especially for the Asterisk GUI project.
Hi,
I'm using queue configuration as follows:
- queues from* queues.conf*
- queue_members from *external Database thru ODBC*, using* Local channels
* as interface
- sip extensions from *external Database thru ODBC*
When a call is sent from queue to an interface (local channel), it
Hi -
OK not really an Asterisk question but it is affecting one of my
favorite features - emailing voice mail! I've posted on some Linux
forums and sendmail.org but no response so I'm hoping someone will
take pity on me ;-)
My ISP requires SMTP authorization and I'm having a heck of a time
Try using SSMTP
http://www.linux.com/articles/132006
It works with any provider for mail sending, and takes 30 seconds to setup.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
-Original Message-
From: [EMAIL PROTECTED]
On Sun, Oct 5, 2008 at 8:04 PM, David Backeberg [EMAIL PROTECTED] wrote:
Isn't IMAP IMAP? Does MS not actually follow the protocol? Why would
it be different?
When I setup my voicemail.conf for IMAP Asterisk does not work right.
sip show peers only shows 1 peer. The CLI is freezing up, etc.
On Tue, Nov 04, 2008 at 04:02:40PM -0600, Jeff LaCoursiere wrote:
Hmm, I wonder if you could run asterisk in a jail? Anyone done that on
FreeBSD for example? That would solve your issues I think. It would
certainly be difficult for your admin to admin asterisk without the CLI.
Depending
On Tue, Nov 04, 2008 at 04:31:58PM -0600, Tilghman Lesher wrote:
On Tuesday 04 November 2008 15:52:10 Ruddy Gbaguidi wrote:
Did you know that any commandyou type in asterisk cli starting with
exclamation point (!) is execute in the shell by asterisk ??
Example :
running
!ls
will run
Hi Rob,
Also try without the r option to the dial command:
http://www.voip-info.org/wiki-Asterisk+cmd+dial
Rob
After I removed the r option I now nearly immediately get the busy tone.
BUT: The wiki says Without this option, Asterisk will generate ring
tones automatically where it is
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