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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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.
--
Leif Madsen
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lable, 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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based on a released major version would not be affected.
+1 to case-sensitivity. It's the right way!(tm)
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conference bridge.
No need to manipulate from the dialplan anymore.
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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
e. Changing this can only make it a backwards
compatibility 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.
--
Leif M
s!!
Oh heck ya. You can 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.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
--
___
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
i
It uses 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 27/09/12 02:13 PM, Mehdi Rahimi wrote:
On Wed, Sep 26, 2012 at 11:31 PM, Leif Madsen
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 music .
Probably Local
ial source
tarball.
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.
which 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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ps://issues.asterisk.org/jira/browse/ASTERISK-20150
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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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Leif Madsen
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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
ns
* 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
--
Leif Madsen
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On 05/01/12 05:24 PM, Kevin P. Fleming wrote:
Although in my personal opinion, it's really
hard to beat the IP5000.
That has been my experience as well.
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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
ctions would be called
dynamically from features.conf.
You'd just create the application_map 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/
iled" 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.
--
Leif Madsen
http://www.oreilly.com/c
you're thinking, but it was never complete, and
has never become a priority issue for any community developer to
complete. You could of course create an AGI() script that pulled the
audio out of the database, caching the audio for a period of time, then
ju
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
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
(rega
On 11-11-09 04:30 PM, Danny Nicholas wrote:
> If you have an "ancient" version of Asterisk you want to stick with, you can
> do this with asterisk -rx "sip set debug on" and asterisk -rx "agi set debug
> on" in your safe_asterisk script.
Not sure about AGI, but pretty sure the sipdebug=yes option
On 11-11-07 08:38 AM, Bryant Zimmerman wrote:
> I have a test box that has been running asterisk without issue. I updated
> it to 1.8.8.0-rc2 and now I am getting some wierd issues I have never seen
> before.
>
>
> All the modules seem to have compiled without issue.
>
>
> Asterisk starts up
On 11-11-10 11:57 AM, Danny Nicholas wrote:
> Misspoke - when should we expect 10.0 that is not -rc or -beta?
Well Asterisk 10.0.0 is now in release candidate status, which means pending any
major issues or regressions, a full Asterisk 10.0.0 is inevitably due in the
near future.
Leif.
--
__
On 11-11-10 11:43 AM, Danny Nicholas wrote:
> Does this mean a "non-beta" labeled Asterisk 10.0 is due out shortly?
I'm confused by shortly the announcement means it is out now. That was the
purpose of the announcement...
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-10.0.0-rc
imply that the 's' extension was a catch-all. In the first and second
editions of Asterisk: The Future 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
m
nge all of my simple dialplans
to read exten=> start,1,blah instead 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 eve
ble you haven't 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.)
--
Leif Madsen
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2/asterisk-Install.html#Installing_id291070
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you define "NAT 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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Leif Madsen
http://www.oreilly.com/catalog/asterisk
--
_
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?
--
Leif Madsen
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along as I thought it was pretty neat when I
learned about it :))
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,NoOp()
same => n,Set(CHANNEL(musicclass)=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.aster
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.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
)
same => n,ExecIf($["${EXTEN:-8}" = "42704701"]?Macro(dialfax,${EXTEN:-8}))
same => n,Verbose(2,Did not match -- falling through)
same => n,Playback(invalid)
same => n,Hangup()
I'm pretty sure that's the only way you can do it in a single line (th
sting 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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Leif Madsen
http://www.oreilly.com/catalog/asterisk
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K, well nothing looks obviously 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.
--
Leif Madsen
http://www.oreilly
came in from 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
http://www.oreilly.com/cata
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.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
log/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 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 var
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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_
-- Bandwidth and Colocation
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 ha
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
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 exper
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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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 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
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 ad
this case, tags/1.8.4-rc3 as copied to tags/1.8.4, and
the only changes were made to the .version file and ChangeLog. Then the standard
release process is followed to turn that tag into a .tar.gz and get it onto the
downloads site.
Any changes made
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
>
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
>&
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
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 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
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?
U
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 co
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 didn't
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 fly-by-ni
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 o
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:
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.
https://reviewboard.asterisk.org/
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 authe
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.
http://downloads.asterisk.org/
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 ca
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 in
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
modifyin
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 Cac
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 fi
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 11-01-24 08:29 AM, RR wrote:
Wow, alright, after an all-nighter, I was able to get timerfd.so compiled in
Asterisk 1.8.2.2 under Debian Lenny 5.0.7 with Kernel 2.6.26-2-amd64. Of
course, due to the glibc requirement of 2.8+, a lot of dodgey upgrades had
to be performed. I have no idea how "sta
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:
http://ofps.oreilly.com/titles/9780596517
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 lik
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 10-12-17 06:48 AM, Gilles wrote:
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
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. Imagine you have an
A
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 s
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 10-11-23 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 10-11-15 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 dia
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.
--
_
-- Bandwidt
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
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 t
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:
>>>
>>> http://downloads.as
On 10-10-15 07:20 AM, Leif Madsen wrote:
> On 10-10-14 10:49 PM, bruce bruce wrote:
>> Unfortunately, probably there is no one you can complain to. But it also
>> sickens me at how badly Asterisk is made to not cope with situations
>> like this and worse than that is FreePBX
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
On 10-10-14 10:49 PM, bruce bruce wrote:
> Unfortunately, probably there is no one you can complain to. But it also
> sickens me at how badly Asterisk is made to not cope with situations
> like this and worse than that is FreePBX.
How is password policy an Asterisk issue? The solution to the probl
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 ext
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
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
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
>
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 f
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