.
No need to manipulate from the dialplan anymore.
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on a released major version would not be affected.
+1 to case-sensitivity. It's the right way!(tm)
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New
, and
we're basically in the same boat as just changing it in the next major
version.
Consistency for the win!(tm)
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to be performed by those affected. Especially, as in the case of what Raj
mentioned at the beginning of his prior email, not too many people may even be
affected by this change just like he won't be.
Well said.
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Leif Madsen
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could give a better idea as to what your needs are, and why
MeetMe() doesn't fulfill them? Perhaps ConfBridge() in Asterisk 10 or
later would fulfill those needs?
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Leif Madsen
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/ASTERISK-20150
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he's using as an example in the upcoming Asterisk: The Definitive
Guide 4e book.
[0] https://github.com/russellb/amiutils
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tarball. snip
That of course also implies contributions to review the files prior to
release (which have release candidates). That directory contains data
that was at one point contributed, and should really be reviewed by the
community with any changes required submitted back upstream.
--
Leif
On 27/09/12 02:13 PM, Mehdi Rahimi wrote:
On Wed, Sep 26, 2012 at 11:31 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 26/09/12 05:35 AM, Mehdi Rahimi wrote:
I want to play music in my AGI while i am searching for a field in DB.
Actually during some processes in AGI i need to play
a modified InnoDB to
allow a multi-master MySQL cluster.
I used a chef cookbook to deploy it[1].
[0] http://www.codership.com/content/using-galera-cluster
[1]
http://support.severalnines.com/entries/21453521-opscode-s-chef-mysql-galera-and-clustercontrol
--
Leif Madsen
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On 28/09/12 07:36 AM, Markus wrote:
Am 28.09.2012 13:24, schrieb Leif Madsen:
Is another channel connected to the conference receiving the DTMF? Is
that what you're intending? Because from my understand that is the
intention, and not simply to limit the DTMF from being in the conference
start up an Asterisk instance and just start doing
things with it via your programs. That's the immense power of AMI; it's
essentially the Asterisk API.
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issue. Someone who has run into this and needs it to act
differently will seek out the new option after reading about it in the
CHANGES file.
In an ideal scenario, a system upgrade should require the least amount
of knob turning.
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Leif Madsen
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On 28/09/12 08:45 AM, Joshua Colp wrote:
Leif Madsen wrote:
I guess part of the question is; can you trigger it to be re-enabled
after the stream file?
Sure you can! You can use set music to start it going again as the next
command.
And that makes sense. I kind of knew the answer already
On 26/09/12 05:35 AM, Mehdi Rahimi wrote:
I want to play music in my AGI while i am searching for a field in DB.
Actually during some processes in AGI i need to play music .
Probably Local channels to the rescue here.
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this in ConfBridge(). Look at
video_mode=follow_talker
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On 27/08/12 10:08 AM, Asterisk Development Team wrote:
As a part of other infrastructure changes we are making
to the community services, we will finally shut down Mantis for good.
Huzzuh!
Does this mean http://issues.asterisk.org will now go directly to JIRA?
Leif.
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http
* usage of SLAtrunk() and SLAstation() applications to use the SLA
lines, which also changes the device status (that is monitored by the
device)
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-AudioManipulation_id302347.html
I also talked a bit about injecting audio onto a channel at AstriCon
2011 in my Cooking With Asterisk talk. It's the last recipe I talk about
in this video: http://www.astricon.net/videos/Cooking-with-Asterisk.html
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Leif Madsen
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as expected.
Also, please be sure to file errata so that we can look at it for the
next printing or version of the book (depending on what the issue
actually is).
Errata can be filed at http://oreilly.com/catalog/9780596517342/errata/
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
On 05/01/12 05:24 PM, Kevin P. Fleming wrote:
snip
Although in my personal opinion, it's really
hard to beat the IP5000.
That has been my experience as well.
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as documented in features.conf and
then apply the PITCH_SHIFT() function to whichever channel you want.
Untested, but should look something like:
pitch_up_them = 3*,peer/both,Set(PITCH_SHIFT(tx)=high)
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Leif Madsen
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of course create an AGI() script that pulled the
audio out of the database, caching the audio for a period of time, then
just played it like you would any other file, and cleaning up the file
after.
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Leif Madsen
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into Asterisk dialplan (like
extensions.conf) anyways, so you can. You just have to follow the same
rules about there not being duplicate macro names, as the AEL and
dialplan logic is going to be combined together in memory.
Leif.
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Leif Madsen
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On 11-11-10 12:12 PM, Danny Nicholas wrote:
Yeah! My boss will be much happier having a system that doesn't have the
-tail on it.
I hear this kind of statement every once in a while, which makes absolutely no
sense to me. If you're blindly running a version of any software in production
On 11-11-10 01:15 PM, Eric Wieling wrote:
The Asterisk source tree has a document with instructions on getting a
backtrace from the segfaults so you can report it on the issue tracker.
Most up to date documentation is on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Debugging
installed an MTA or have it disabled. Or
perhaps the other ends are rejecting due a missing MX record, or some
other email configuration issue.)
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of exten = s,1,blah. To me exten=
s,1,blah is more intuitive and less vulnerable than exten = _X.,1,blah.
The 's' extension does stand for 'start' but I don't think we've ever
implied it was a catch-all extension.
--
Leif Madsen
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of Telephony we were mostly using
analog lines, and thus the usage of the extension 's' was fairly
prominent. There are many other single letter extensions that have extra
meaning, such as 'i', 't', etc..., but we never intended to imply that
's' was a catch-all extension.
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Leif Madsen
http
problem? I'm unaware of any issues with Asterisk
(or end points) behind NAT. It is mostly likely a configuration issue
rather than a bug.
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-Install.html#Installing_id291070
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On 08/09/11 02:19 PM, Cobra 2 wrote:
I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and
I've gotten asterisk to run on that just fine.
I think the question is, can you answer your incoming calls with the
Asterisk running on the device?
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On 04/09/11 02:51 PM, Tamer Higazi wrote:
the 3rd edition is available, but that book covers every thing to run
the asterisk PBX.
You can read the 3rd edition online at
http://ofps.oreilly.com/titles/9780596517342/
HTH!
Leif.
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Leif Madsen
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)=default)
I could use just:
exten = s,n,MusicOnHold()
There is a lot of documentation on www.voip-info.org but sometimes it is
not clear which asterisk version it applies to :-/
(Another good reason to be reading the documentation on
https://wiki.asterisk.org/wiki instead :))
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Leif Madsen
it was pretty neat when I
learned about it :))
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= n,Playback(invalid)
same = n,Hangup()
I'm pretty sure that's the only way you can do it in a single line (the ExecIf()
application).
Leif Madsen.
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New
changes
in your development systems at any period of time.
1.8.5 release candidates should be available later this week though.
Leif.
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the ITSP. If that channel is then transferred, the
recording should follow it around.
Can you elaborate a bit more on the call flow and show the console output?
--
Leif Madsen
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wrong there from what I can tell.
What is your console output doing though when you do the transfer? Are
you using Asterisk transfers? What version of Asterisk are you using?
Leif.
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Leif Madsen
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On 02/06/11 03:35 PM, satish patel wrote:
Is this available in current SVN ?
Changes are always checked into SVN first and then made available in a tag.
Leif.
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Leif Madsen
http://www.oreilly.com/catalog/asterisk
/9780596517342 (left hand side). That way we
can get it fixed up in subversion.
Thanks!
Leif.
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New
-ACD.html#ACD_id288626
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On 11-05-24 01:21 PM, Steve Edwards wrote:
If it take the OP (of this thread) 3 years to reply, what does that say about
their product support?
Par for the course? :)
Leif.
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On 11-05-20 09:37 AM, Ishfaq Malik wrote:
Do many people use this?
Is it reliable and safe?
It may still work, but that code is quite old, and I'm not even sure it's
necessary any more.
Leif.
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On 11-05-20 10:39 AM, Benoit Panizzon wrote:
After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is
just put in a temporary variable __SIPDIVERSIONREASON but not in a variable
useable in the dialplan.
You could double check by using DumpChan() to see what channel
On 11-05-16 07:29 AM, Olivier wrote:
As this bug is considered fixed, I think you can't add any comment
anymore.
Unfortunately, you can still see lines mentionning DEVSTATE function like :
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, DEVSTATE function called with no
On 11-05-16 09:13 AM, Alex Balashov wrote:
On 05/16/2011 09:00 AM, Mohammad Khan wrote:
Is there way I can use two Asterisk box, one to maintain SIP packets and
other for RTP traffic?
No, the signaling and bearer plane are integrated in Asterisk.
But you can use reinvites to hand off
On 11-05-13 11:39 AM, isr...@gmail.com wrote:
I haven't tried with timerfd but with timer pthread 1.8 is very unstable
I think I have seen a post to the list from kevin fleming that the same is
for timerfd that there is a nasty bug which they haven't found the reason for
yet
My
On 11-05-11 06:36 PM, Skyler wrote:
Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd
like to take a look at it for sure. The dial plan example Leif replied with
is pretty much what I was thinking, just didn't have a clue how to go about
it. ;)
You could also look
On 11-05-11 09:31 PM, Jose P. Espinal wrote:
Download links on the website have not been updated (asterisk.org)
Oops sorry! I will fix that right.. now!
Leif.
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and get it onto the
downloads site.
Any changes made after 1.8.4-rc1 (for example) would then become available in
1.8.5-rc1, because only RC1s contain all changes from the branch directly.
HTH,
Leif Madsen.
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On 11-05-11 12:57 PM, Skyler wrote:
I would like to track/store concurrent call usage per user by
day/week/month and get server totals by day/week/month. Google comes up with
mostly info regarding concurrent call limits, though my goal is to calculate
actual concurrent channel usage and add
On 11-05-11 12:29 PM, Steve Edwards wrote:
On Wed, 11 May 2011, Eric Wieling wrote:
Generally you should insert a Noop in the dialplan to examine variables.
Noop(EXTEN is ${EXTEN}) for example.
The 'verbose()' application would be an example of 'better practices.'
It's function is
On 11-05-06 02:56 PM, Watkins, Bradley wrote:
Yes, use the MinivmMWI application.
That's how I've done it in the past as well.
Leif.
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On 11-04-30 03:10 PM, Alec Taylor wrote:
Good Evening,
I'm setting up an Internet Radio website with call-in functionality,
and need to know the kinds of FOSS tools I should install to get the
job done.
Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png
Call
On 11-04-29 02:59 AM, Olle E. Johansson wrote:
29 apr 2011 kl. 01.49 skrev Leif Madsen:
Well the issue is that we currently have over 900 open issues in the Asterisk
project alone, and with only one primary bug marshal (myself) sometimes
things
accidentally get closed if it looks like
On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for
few weeks/monthes till 1.8 reaches the level that the community accept to
switch
to 1.8
What is the guide here? What is the level that the community accepts?
On 11-04-28 04:33 PM, Administrator TOOTAI wrote:
Le 28/04/2011 21:47, Leif Madsen a écrit :
On 11-04-28 12:04 PM, Administrator TOOTAI wrote:
Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes
for
few weeks/monthes till 1.8 reaches the level that the community accept
On 11-04-28 07:02 PM, Ira wrote:
At 03:48 PM 4/28/2011, you wrote:
OK, maybe not, but if I thought it was a bug and you discover it was a bug and
fix it, than who was it who decided it wasn't a bug 15 minutes after I put it
in
the bug tracker and why did that person have that much power?
On 11-04-28 07:09 PM, Alec Davis wrote:
Making an assumption here, I'm sure I cleared the remaining resequencing
issues up in 1.4 SVN and 1.6.2 SVN.
https://issues.asterisk.org/view.php?id=19032
The issues I uncovered and fixed were when a new voicemail is left, while a
mailbox is open for
On 11-04-18 02:47 PM, Jerry Geis wrote:
When I do core show channels concise over the AMI interface
how do I specify that I want to see the actual channel number like
DAHDI/4/xxx
where 4 is the actual channel.
RIght now I am seeing DAHDI/i1/x where i1 is the span.
I could have
On 11-04-07 09:45 AM, --[ UxBoD ]-- wrote:
And don't forget that call pickup crashes Asterisk from what would appear
release 1.8.1 upwards! We have had to back level to that latest 1.6 branch.
https://issues.asterisk.org/view.php?id=18654
I ran into this issue as well on 1.8.3.2, but I
On 11-03-03 11:22 AM, Brent A. Torrenga wrote:
I am becoming frustrated with our current VOIP provider. Does anyone have
any suggestions for a provider that supports asterisk well and provides
solid service? Voip-info.org has a husge list of providers, but it is
impossible to tell the
On 11-02-27 09:12 PM, Stuart Longland wrote:
I've tried researching this, and so far, have struggled to find any
contemporary information on the issue, so I do apologise if asking this
irritates people who have answered this before.
I have managed to set up Asterisk 1.8 on the web server here.
On 11-02-24 08:56 PM, Andrew Latham wrote:
And I go back to triple check and compare revision numbers... You are
100% correct, the revision numbers in our local repository are wrong,
someone pushed the 1.8.3 RC3 into our local 1.8.2 branch. I apologize
and will work to better control my trust
On 11-02-24 04:08 PM, Andrew Latham wrote:
There are many updates in 1.8.2.4 that may fix your issue. If you are
running any version of 1.8 it should be a quick update.
I wouldn't say many. There is one fix in 1.8.2.4 over 1.8.2.3.
From the ChangeLog:
* Asterisk 1.8.2.4 Released.
On 11-02-23 10:31 AM, Jose P. Espinal wrote:
-
Added a new configuration option remotesecret for authentication to
remote services. For backwards compatibility, secret still has the
same function as before, but now you can configure both a remote secret
and a local secret for mutual
On 11-02-23 10:31 AM, Jose P. Espinal wrote:
Hello List,
I have a little issue with calls placed to a provider declared on sip.conf,
because of a not clear (*for me*) behavior of 'remotesecret' parameter.
Actually I was wrong!
See here. It is being resolved.
On 11-02-22 10:16 AM, Ishfaq Malik wrote:
Has this issue been fixed in this release of 1.8 (or even in the
previous 1.8.2.3)?
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
No. The ChangeLog would give you the information you're looking for.
On 11-02-14 05:10 PM, Kevin P. Fleming wrote:
On 02/14/2011 04:08 PM, Jian Gao wrote:
I am building a server for a client. I want them to try out the new
Google Voice feature using my GV account. But I don't want expose my
GV's password.
There is no method to obscure a Google Voice password
On 11-02-14 05:08 PM, Jian Gao wrote:
I am building a server for a client. I want them to try out the new Google Voice
feature using my GV account. But I don't want expose my GV's password.
Actually in this case, your best bet is just going to be to create a separate
account where you don't
On 11-02-13 09:52 AM, Gilles wrote:
I'm using Asterisk 1.4.20, and can't have Asterisk not load modules I
don't need:
Does someone know why Asterisk still loads modules even with the above
lines in modules.conf?
It looks like you're loading Asterisk, which loads all the modules, then
On 11-02-01 05:22 PM, Juan David Diaz wrote:
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI or a SIP trunk with the following configuration
Dell R710
Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single
Intel® Xeon® X5650, 2.66Ghz, 12M
On 11-01-26 08:52 AM, Gilles wrote:
Hello
I'd like to display CID information on users' monitor running
Windows.
You could use any XMPP client and send a message to it using JabberSend() from
the dialplan. We document using it at http://ofps.oreilly.com.
Leif.
--
On 11-01-26 04:07 PM, Kevin P. Fleming wrote:
On 01/26/2011 03:06 PM, Warren Selby wrote:
Just curious, but why is this 1.8.2.3 and not just 1.8.3? I thought the new
versioning methods made updates into 1.8.x releases and security updates into
1.8.x.y releases?
Security fixes and regression
On 11-01-23 02:56 PM, Jeff B wrote:
There does not seem to be very much info out there about using LDAP to
create asterisk configurations. Does anyone have some information
that they would suggest I start with?
We've tried to document some of it here:
On 11-01-21 08:52 AM, Andrew Thomas wrote:
I know that the 'fix' has just been applied
(https://issues.asterisk.org/view.php?id=18262) - but why does it stop
the moh only to start it again? This, also, seems to cause a CDR
problem (see below).
After speaking with Shaun and Russell, this is
On 10-12-17 06:48 AM, Gilles wrote:
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone.
On 10-12-17 06:17 AM, Olivier wrote:
Hi,
What is currently missing in Asterisk ecosystem to get 2 servers active-active
redundancy such as when server 1 is failing (in some circumstances), its ongoing
calls (or most of them) are kept alive and handed over to server 2 ?
I remember that a couple
On 10-12-15 09:46 AM, bilal ghayyad wrote:
Hi All;
I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8?
For example, when to decide that I have to go for 1.6 or I have to go for 1.8?
It depends on your required usage (features available in version) and your
required
On 10-11-23 07:31 AM, --[ UxBoD ]-- wrote:
I have read the wiki entry but unsure when we would likely see a 1.8.0.1 beta
or release candidate ?
It will be Asterisk 1.8.1-rc1 and that is now available (as of a few minutes
ago)
http://www.asterisk.org/node/51466
Leif.
--
On 10-11-23 08:24 AM, Henry Dogger wrote:
I have an aastra 6739i which supports the g722 codec.
Which format setting do I need to be able to record in wideband?
Tried: wav, gsm, pcm. Nothing seems to give me the result I desire.
Shouldn't you try g722 as the format?
Leif.
--
On 10-11-15 08:30 AM, Richard Kenner wrote:
It's kind of low for me. How does one control that volume?
You could use the VOLUME() function prior to joining the conference for
channels
that are quiet.
Leif.
--
_
--
On 10-11-15 06:04 PM, Matt Darnell wrote:
Is this command the best way to access a MySQL database -
MYSQL(Connect connid dhhost dbuser dbpass dbname) ?
I thought I heard that using ODBC was a bit more stable.
Anyone have any experience?
Use func_odbc along with res_odbc. I've taken
On 10-10-25 04:21 PM, Dan Journo wrote:
Hi,
When a VOIP user dials an external number, the calls are routed through
our SIP provider.
Is there a simple way to check whether the DDI exists locally before
dialling out to the sip provider?
Something like GotoIfExists(5551...@incoming_calls)
On 10-10-18 11:01 PM, Barry Miller wrote:
On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote:
On 10-10-18 07:54 PM, Asterisk Development Team wrote:
For a full list of changes in the current release candidate, please see the
ChangeLog:
On 10-10-19 10:46 AM, Danny Nicholas wrote:
Greeting list,
I hope this isn’t a “lazy” question. I have been running TDM400P and
TDM410P cards in Dell PowerEdge Servers for a few years now. We are
moving from physical servers to VMWARE servers. What opportunities
should I expect moving these
On 10-10-15 04:10 AM, Сикорский Сергей wrote:
15.10.2010 9:40, Warren Selby пишет:
I think this means you need to set a call-limit for each sip peer
Is there any alternative for obsolete call-limit option in 1.6/1.8?
The correct answer is to use ringinuse=no in queues.conf and callcounter=yes
On 10-10-14 12:18 PM, Carlos Chavez wrote:
I have a customer that has a Trendnet TEW-435BRM router which has the
bad habit of rewriting all external connections so the Asterisk server
only sees the IP address of the router itself. Up to today this has not
been a problem since all
On 10-10-04 09:44 AM, Flavio Miranda wrote:
Hi all,
Every time I reload my asterisk it fall down and the following message
appear on log:
parse error: No category context for line 7 of /etc/asterisk/chan_dahdi.conf
If I comment that line, it change to other line.
There are some thing
On 10-10-04 10:59 AM, Flavio Miranda wrote:
Asterisk:/var/log/asterisk# pico /etc/asterisk/chan_dahdi.conf
; DAHDI telephony
;language=en
;echocancel=yes
echocancelwhenbridged=yes
ss7type = itu
ss7_called_nai=dynamic
ss7_calling_nai=dynamic
;General options
usecallerid = yes
On 10-09-26 01:00 PM, bilal ghayyad wrote:
First of all, I am looking to have the H323 Gatekeeper service available at
Asterisk, and really I do not know if 1.4 or 1.6 or 1.8 started implementing
H323 gatekeeper functionality or not?
Until 1.4.26.2 version, there is no h323 gatekeeper
On 10-09-26 02:55 PM, Ira wrote:
At 10:37 PM 9/24/2010, you wrote:
You probably need to install libssl-dev then rerun ./configure. At
least I did (Debian Lenny). Seems chan_sip needs res_crypto which
needs libssl.
Thanks, I tried to figure out what I needed but I failed. That was
it,
On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote:
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and when I hit mute on the
softphone, all rtp traffic
On 10-09-23 07:01 PM, Mike wrote:
Hi,
I have a server with multiple IP address, Asterisk binding with all of
them. I'd like Asterisk to reply to a SIP peer from the same IP address
as the peer used to register to Asterisk (as opposed to using the main
IP address all the time regardless of
On 10-09-22 11:45 AM, Klaus Darilion wrote:
Hi!
Since some time the download of the newest Asterisk does not contains
the version number anymore, but is just called asterisk-1.4-current.tar.gz
This gives me a tarball where I do not know the version without looking
into the tarball.
Thus,
On 10-09-16 09:43 AM, Dan Journo wrote:
That's not a bug. Only when the phone registers or performs some sort of
action
(such as placing a call, etc...) does Asterisk query the database. If your
phones have a short re-registration time this becomes less of a problem.
How do you explain that
On 10-09-15 05:25 AM, Jonas Kellens wrote:
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx formule) in new stack
[Sep 15 11:16:32] -- Executing
On 10-09-15 12:13 PM, Paul Belanger wrote:
On Wed, Sep 15, 2010 at 11:54 AM, Ryan Wagonerrswago...@gmail.com wrote:
Anybody else notice that the 1.6.2.12 download has a version and
changelog for 1.6.2.12-rc1?
I can confirm, asterisk-dev notified.
Odd, not sure how this happened, but I'll be
On 10-09-15 03:41 PM, Dan Journo wrote:
I think ive found a bug but need someone to double check.
Whenever I issue a reload in Asterisk, any realtime extensions stop
receiving calls.
I have to reboot the sip phones in order to get them to re-register.
Can anyone see if they have a similar
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