How many people do you plan to page? because if numbers are high (or
variable) you may have an easier life by using some sort of dialer if
numbers are not very high and two lines are enough, our WombatDialer is
free to use.
l.
2012/12/29 bilal ghayyad bilmar...@yahoo.com
2) Praying time
With just one PRI card this should not be an issue, but for larger systems
you may consider using something like Oreka to offload the I/O from the
Asterisk server
l.
2012/12/31 Vinod Nadiadwala thinw...@gmail.com
Hi,
I am new to asterisk, i want to know that is it possible to use
Steve Murphy submitted a patch a while ago to track MOH on queues, you can
find it at https://issues.asterisk.org/jira/browse/ASTERISK-20742 - it
could be a good starting point to work on as it is quite short.
Too bad it is still in limbo :-(
l.
2012/12/19 Andrew White
So where has every body else gone? :)
l.
2012/12/30 Mr. James W. Laferriere bab...@baby-dragons.com
2003, 24471
2004, 48608
2005, 59116
2006, 41215
2007, 26414
2008, 20746
2009, 18304
2010, 14948
2011, 11588
2012, 7542
--
Top post for the New Year.
Yes, if you might scale up to 60 or more simultaneous calls,
definitely look at OrecX or RTPTap because you will run into I/O
issues. Not sure what current hardware can accommodate but it is best
not to find out.
Considering the very low cost of hardware these days
I don't know how many I/O can be achieved on a modern hardware, but I don't
think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of
data. However can be a good idea to start loading a server and be prepared
to share the load on another server.
Leandro
2013/1/2 Steve Totaro
So where has every body else gone?
Still here, but mature working systems, still running 1.4.x
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
Doesn’t the OP wish to page all phones? So it’s not an issue of dumping dozens
of call files all at once.
Does paging work?
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page
http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom
Overhead paging might also be something to
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,…)
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
To: asterisk-users@lists.digium.com
Subject:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Wednesday, January 02, 2013 7:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Users list email totals by year .
So
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
Bob R
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New to Asterisk? Join us for a live introductory
It depends on what you do with them.
Years ago, 60 calls would start to crap out audio on live calls and I
learned that the hard way on a production call center. There was the
I/O of just SLIN, then converting to MP3, then transferring to a not
too forgiving SAMBA share. Scheduling things for a
On 13-01-02 10:55 AM, Robert Rawlinson wrote:
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
Bob R
If the Pi is running debian or a variant thereof, won't
# apt-get install asterisk
work?
--
Looking for (employment|contract) work in the
Mixmonitor also muxes the two sides of the conversation after hangup.
That is quite a bit of I/O for 60 simultaneous calls lasting an
average of 5-15mins
On Wed, Jan 2, 2013 at 9:59 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
It depends on what you do with them.
Years ago, 60 calls
On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote:
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
apt-get install asterisk
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
On Wed, 2 Jan 2013, Robert Rawlinson wrote:
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
Maybe you will find this interesting:
http://nerdvittles.com/?p=3880
Regards,
--
Tom m...@tdiehl.org Spamtrap address
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and we
announce the call before the transfer is complete. The carrier requires a
diversion header if the ANI is not one of our DIDs. Does someone have
experience
Henrik Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension
On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dkwrote:
Gergo Csibra csi...@gmail.com writes:
Complaining about top posting on a list where's no moderation,
no sanction if somebody top posting is pointless.
There is a sanction. People like me will score top posters
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Harrington
Sent: Wednesday, January 02, 2013 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Top
On 2 Jan 2013, at 15:54, Eric Wieling wrote:
On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dk
wrote:
There is a sanction. People like me will score top posters lower and
soon not see their posts at all.
I'm the opposite. I'm likely not to scroll down 10 pages to
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, January 02, 2013 9:54 AM
To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
I'm the opposite. I'm likely not to scroll down 10 pages to see
the comments at the end.
Wouldn't need to if people trimmed their posts properly.
Precisely (e.g., see above)! Indeed, my sense is that top-posting
*discourages* properly trimming email and that's my main reason against it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Wednesday, January 02, 2013 10:00 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Top Posting
I'm the opposite. I'm
On Wed, Jan 2, 2013 at 9:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote:
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
apt-get install asterisk
Does anyone know of any
On 2 January 2013 16:16, Chris Gentle gent...@gmail.com wrote:
Does anyone know of any asterisk 11 packages for the Pi? I ended up
compiling it myself this weekend. Took a while.
Take a look at http://www.raspberry-asterisk.org/ :)
--
I'm the opposite. I'm likely not to scroll down 10 pages to see the
comments at the end.
Wouldn't need to if people trimmed their posts properly.
Precisely (e.g., see above)! Indeed, my sense is that top-posting
*discourages* properly trimming email and that's my main reason against it.
On Wed, Jan 2, 2013 at 11:00 AM, Richard Kenner ken...@gnat.com wrote:
I'm the opposite. I'm likely not to scroll down 10 pages to see
the comments at the end.
Wouldn't need to if people trimmed their posts properly.
Precisely (e.g., see above)! Indeed, my sense is that top-posting
On Wed, Jan 2, 2013 at 10:19 AM, Dan Jenkins
dan.jenk...@holidayextras.comwrote:
On 2 January 2013 16:16, Chris Gentle gent...@gmail.com wrote:
Does anyone know of any asterisk 11 packages for the Pi? I ended up
compiling it myself this weekend. Took a while.
Take a look at
In this properly trimmed example, there's no record of who said what.
When it's relevant, I trim in such a way that that information is
preserved. But I would *never* leave in a header, just the identification
of the person who typed that part. Most mailers, when you include text
from another
If things were properly trimmed, the email would be short enough that it
really doesn't matter that much if the new material is on the top or
bottom, but people who top-post and don't trim create really hard-to-follow
emails.
Not really true often times when people do the right thing
I finally got it to happen again.
#0 0x00296f96 in __memcpy_ia32 () from /lib/libc.so.6
#1 0x0002 in ?? ()
#2 0x4d44fa0e in snd_pcm_area_copy () from /usr/lib/libasound.so.2
#3 0x4d44ff09 in snd_pcm_areas_copy () from /usr/lib/libasound.so.2
#4 0x4d4620f4 in snd_pcm_mmap_read_areas ()
On 1/2/2013 11:30 AM, Richard Kenner wrote:
If things were properly trimmed, the email would be short enough that it
really doesn't matter that much if the new material is on the top or
bottom, but people who top-post and don't trim create really hard-to-follow
emails.
Not really true often
What version of ALSA do you have installed? 1.0.26 is current (
http://alsa-project.org/main/index.php/Main_Page ) and it looks like the
crash is in there.
On Wed, Jan 2, 2013 at 10:32 AM, Jerry Geis ge...@pagestation.com wrote:
I finally got it to happen again.
#0 0x00296f96 in
On 1/2/2013 12:00 PM, j...@millican.us wrote:
On 1/2/2013 11:30 AM, Richard Kenner wrote:
If things were properly trimmed, the email would be short enough
that it
really doesn't matter that much if the new material is on the top or
bottom, but people who top-post and don't trim create really
Grow up, follow the rules, have a good day.
JohnM
PS. Did not intend to imply that it was Steve that hijacked the thread, in
case anyone read my comment that way JohnM
Steve has waded through enough of these that he should be a hijacker.
--
What version of ALSA do you have installed? 1.0.26 is current (
http://alsa-project.org/main/index.php/Main_Page ) and it looks like the
crash is in there.
I am using 1.0.25
Jerry
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-- Bandwidth and Colocation Provided by
On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote:
On 1/2/2013 11:30 AM, Richard Kenner wrote:
If things were properly trimmed, the email would be short enough that it
really doesn't matter that much if the new material is on the top or
bottom, but people who top-post
On 1/2/2013 12:20 PM, Steve Totaro wrote:
On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote:
On 1/2/2013 11:30 AM, Richard Kenner wrote:
If things were properly trimmed, the email would be short enough that it
really doesn't matter that much if the new material is on the
On 01/02/2013 12:20 PM, Steve Totaro wrote:
good one - me too !
On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote:
On 1/2/2013 11:30 AM, Richard Kenner wrote:
If things were properly trimmed, the email would be short enough that it
really doesn't matter that much if
At 08:22 AM 1/2/2013, you wrote:
Wouldn't need to if people trimmed their posts properly.
Precisely (e.g., see above)! Indeed, my sense is that top-posting
*discourages* properly trimming email and that's my main reason against it.
If things were properly trimmed, the email would be short
It is not hard to follow the rules . If the nice folks at Digium took the
time to post rules we should at least TRY to follow them. If you do not like
the rules you can always petition Digium to change them but, taking up
bandwidth on the list in this all to frequent pissing match is a futile
On Wed, Jan 2, 2013 at 11:02 AM, Ira i...@extrasensory.com wrote:
And I started communicating with a 2400 baud modem so trimming was a
necessity and a requirement of friendship.
Bah, spoiled kids. Mine was a 110 baud acoustic.
I think the Will Asterisk run on a Rasberry Pi thread the
On 1/2/2013 1:10 PM, Don Kelly wrote:
It is not hard to follow the rules . If the nice folks at Digium took the
time to post rules we should at least TRY to follow them. If you do not like
the rules you can always petition Digium to change them but, taking up
bandwidth on the list in this all
On 02/01/2013 1:11 PM, Carlos Alvarez wrote:
On Wed, Jan 2, 2013 at 11:02 AM, Ira i...@extrasensory.com
mailto:i...@extrasensory.com wrote:
And I started communicating with a 2400 baud modem so trimming was
a necessity and a requirement of friendship.
Bah, spoiled kids. Mine was a
On 01/02/2013 07:11 PM, Carlos Alvarez wrote:
The number of questions posted here that are easily answered with a
search or which are far too basic and open (how do I make Asterisk work)
is very high these days, and that does kill a list. A lot of us are
interested in helping people who help
On 1/2/2013 Don Kelly wrote:
... what product/procedure/whatever would
enable me to follow and participate in bottom-posted discussions as it
doesn't appear that Outlook or gmail are very effective.
Umm, what about positioning the cursor below the previous post before
writing your reply in
It would be nice (for me anyway) if the mailing list and forum were
combined. Google Groups does this nicely I believe.
Mitch
On 01/02/2013 08:53 AM, Eric Wieling wrote:
I don't use forums as my web browser can't automatically filter the messages
for me like my e-mail program can.
I
On 01/02/2013 06:20 PM, Steve Totaro wrote:
I became a list member way before any such rule and never had to click
through and agree to these update ToS.
I am grandfathered in.
Just looked it up. I see my first post back in April 2003, yours in
September 2003 and Jon in March 2003. Wow you
I have connected a PSTN line to a Digium FXO card.
There is also an ordinary analogue phone attached to the same line.
The Asterisk answers the line on the first ring.
I would like it to wait for a few seconds so that someone can answer the
PSTN line with an analogue phone.
This would allow a
On Wed, Jan 2, 2013 at 12:25 PM, j...@millican.us j...@millican.us wrote:
On 1/2/2013 12:20 PM, Steve Totaro wrote:
On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us
wrote:
On 1/2/2013 11:30 AM, Richard Kenner wrote:
If things were properly trimmed, the email would be
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Wednesday, January 02, 2013 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk as answering
Problematic at best. Just make a phone an extension and allow that to
ring in a hunt group.
Thanks,
Steve Totaro
On Wed, Jan 2, 2013 at 3:30 PM, Ron Wheeler
rwhee...@artifact-software.com wrote:
I have connected a PSTN line to a Digium FXO card.
There is also an ordinary analogue phone
I recommend using WaitForRing instead of Wait.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, January 02, 2013 3:33 PM
To: rwhee...@artifact-software.com; 'Asterisk Users
Hi,
one more hint... (trying to translate the commands to english)
in Thunderbird open - Extras - Filter.. -
Filter-Name: enter Top Posting
Subject - Contains: enter Top Posting
Action: Delete
Markus
Am 02.01.2013 21:31, schrieb Steve Totaro:
On Wed, Jan 2, 2013 at 12:25 PM,
On 01/02/2013 12:16 PM, Don Kelly wrote:
I don't think Outlook does what I'd like, so I'm not limiting my options. I
can use different email to keep track of the Asterisk lists.
Thunderbird (by default) bottom posts. And it does the nice indenting
and allows you to turn off that HTML crap...
On 01/02/2013 03:22 PM, Patrick Lists wrote:
On 01/02/2013 06:20 PM, Steve Totaro wrote:
I became a list member way before any such rule and never had to click
through and agree to these update ToS.
I am grandfathered in.
Just looked it up. I see my first post back in April 2003, yours in
On Wed, Jan 2, 2013 at 3:46 PM, jon pounder j...@inline.net wrote:
On 01/02/2013 03:22 PM, Patrick Lists wrote:
On 01/02/2013 06:20 PM, Steve Totaro wrote:
I became a list member way before any such rule and never had to click
through and agree to these update ToS.
I am grandfathered in.
On 01/02/2013 03:35 PM, Jim Lucas wrote:
On 01/02/2013 12:16 PM, Don Kelly wrote:
I don't think Outlook does what I'd like, so I'm not limiting my
options. I
can use different email to keep track of the Asterisk lists.
Thunderbird (by default) bottom posts. And it does the nice indenting
On 02/01/2013 3:33 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Wednesday, January 02, 2013 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
attachment was scrubbed...
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is taken care of. #2 I'm supposing that you could do a SIP
Header command before the Dial to resolve the diversion header issue.
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43b1f
Asterisk Project Security Advisory - AST-2012-014
ProductAsterisk
SummaryCrashes due to large stack allocations when using
TCP
Asterisk Project Security Advisory - AST-2012-015
ProductAsterisk
SummaryDenial of Service Through Exploitation of Device
State Caching
/attachments/20130102/
459
43b1f/attachment-0001.htm
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Asterisk? Join us for a live introductory webinar every Thurs
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones,
and 11.1.1.
These releases are available for immediate
Gmail has just updated some stuff and I've been fiddling with the gmail ap
on Android (Ice Cream Sandwich).
I can select inline reply, delete superfluous stuff and go to the bottom
for my post.
After a few messages back and forth, the thread is displayed with a Show
quoted text link for each
On 01/02/2013 09:46 PM, jon pounder wrote:
On 01/02/2013 03:22 PM, Patrick Lists wrote:
On 01/02/2013 06:20 PM, Steve Totaro wrote:
I became a list member way before any such rule and never had to click
through and agree to these update ToS.
I am grandfathered in.
Just looked it up. I see
Happy New Year.
Digium Asterisk World is a set of conference sessions that run at IT Expo in
Miami Beach, FL at the end of this month.
See http://www.tmcnet.com/voip/conference/digium-asterisk-world/default.htm
for more details.
The conference runs 1/30-2/1, and we have a small number (2 or
Greetings all,
I have been seeing a lot of
[Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite:
Sending fake auth rejection for device
100sip:100@108.161.145.18;tag=2e921697
in my logs lately. Is there a way to automatically ban IP address from
attackers within asterisk
On Wed, Jan 2, 2013 at 3:49 PM, Frank fr...@efirehouse.com wrote:
Greetings all,
I have been seeing a lot of
[Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite:
Sending fake auth rejection for device 100sip:100@108.161.145.18;**
tag=2e921697
in my logs lately. Is
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Wednesday, January 02, 2013 4:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto ban IP addresses
On Wed, Jan 2,
Hi,
Fail2ban
http://en.gentoo-wiki.com/wiki/HOWTO_fail2ban
-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Frank
Enviada em: quarta-feira, 2 de janeiro de 2013 20:50
Para: Asterisk Users Mailing List -
Howto fail2ban in asterisk
http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Frank
Enviada em: quarta-feira, 2 de janeiro de 2013 20:50
OK. I'm getting out the fireproof suit because it's coming and my
hackles have been raised by a number of comments on the list of late.
Disclaimer:
No disrespect intended to the individuals of any *specific* thread. I'm
a little frustrated over energy wasted on pedantic top/bottom posting
Hi;
How can I know the duration that the DAHDI channel is still used? I need to
know its status and since when it is in this status, how?
Also, is it possible to hangup the channel if it has been openned more than 90
minute? Other than using the timeout in the Dial command (because this I know
I've created some images. I currently don't have a free Raspberry Pi so I
have not updated any images for a little while.
A how to on building your own.
www.klaverstyn.com.au/david/wiki/index.php?title=Asterisk_for_Raspberry_Pi
A how to on writing a pre-compiled image
Thanks for the help.
As I see that the call file is used to generate calls, can I use this technique
to page the Phones?
It is one wave file only that need to be Paged for all the Phones connected on
the Asterisk PBX.
When I say Paging, I mean that they are going to hear the sound from the
Please trim cruft irrelevant to the current questions.
On Wed, 2 Jan 2013, bilal ghayyad wrote:
As I see that the call file is used to generate calls, can I use this
technique to page the Phones?
Yes. The call file would look something like:
application:page
data:
Hi,
I setup PBX with A400P 4 x FXo board. There are one analog line plugged
into port 1.
Internal extension cane make calls to PSTN without any issue.
When I make inbound call, caller get busy tone user busy' message right
away.
Asterisk log shows following log and internal extension (200)
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