El 11/06/2014 1:52 p. m., Matthew Jordan escribió:
On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington w...@willwh.com
mailto:w...@willwh.com wrote:
Chrome 35 broke all of this you need to be using DTLS now I
believe.
I had working secure web sockets with asterisk 12.2.x
El 27/04/2014 8:39 p. m., Sean Darcy escribió:
On 11.9.0:
-- Accepting AUTHENTICATED call from 111.xxx.yyy.zzz:
-- requested format = speex,
-- requested prefs = (),
-- actual format = ulaw,
-- host prefs = (silk16|ulaw|gsm|g722),
--
Known as Virtual Hold, you'll have to program inside asterisk to achieve
that.
El 31/05/12 10:48, equis software escribió:
Is there any option in Asterisk distribution of this?
Thanks.
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El 08/08/11 11:46, J Gao escribió:
On 11-08-06 10:06 AM, Miguel Molina wrote:
El 05/08/11 13:20, J Gao escribió:
I am using the new 1.8.5 and I just found out that Asterisk won't
record the call if the call just hangup. I did a test like this:
exten = 1009, 1, Hangup()
Then I called 1009
El 05/08/11 13:20, J Gao escribió:
I am using the new 1.8.5 and I just found out that Asterisk won't
record the call if the call just hangup. I did a test like this:
exten = 1009, 1, Hangup()
Then I called 1009:
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing
El 18/07/11 18:03, bilal ghayyad escribió:
Dears;
If I need to login using as agent using the AddQueueMember(team,) then what
to be the second paramter? How to be written?
For example, if the agent id is 8000 then it will be:
AddQueueMember(CustomerSupport,Agent/8000) or something else?
El 08/07/11 12:50, Steve Edwards escribió:
*) You can execute hundreds of AGIs written in C in the time it takes
to load the Perl interpreter and parse your script.
Just curious... have you timed this to demonstrate?
Hi,
The VoiceMail() application is to leave messages in mailboxes, to enable
users to access their voicemails, check the application VoiceMailMain().
core show application VoiceMailMain
Regards,
El 09/06/11 17:57, motty.cruz escribió:
Hello, I'm new to this list. I'm trying to configure my
'
Therefore, the facilite is not working!!
What I am doing wrong, could somebody point me out please?!!
Thanks in advanced!!
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than a cellphone! ;-)
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... (but then of course I can't use that account for outbound
calls..)
Adrian
If you use host = dynamic, I think the device must register with
Asterisk for incoming calls go to the right context.
Regards,
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found a similar
MFC/R2 configuration in this post:
http://lists.digium.com/pipermail/asterisk-r2/2010-April/001760.html
Hope it helps.
Cheers,
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, not in the response of the action.
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New to Asterisk? Join us for a live introductory webinar
be used with asterisk to stream say, gsm or
wav files for MoH? I'd appreciate this info, we usually use 'files' mode
and changing that could lower the load on asterisk.
Thanks,
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I think the asterisk team wants, that is,
to focus on only one well supported version instead of having to support
several parallel branches which mean more work and cross-fixing
between them.
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You didn't attach some debug output that shows some work, and you didn't
even tell us what asterisk version are you using, which scenario is on, etc.
Don't expect people to run and answer right away with an inmediate
solution to this.
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it
should be good for you.
(Did I mention I'm not a lawyer?)
I think the OP is asking for the old MoH sound (fpm-world-mix, etc) that
came with asterisk. I wonder why the change from the fpm sounds to the
opsound ones, it was a licensing issue?
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for 1.4.
The good old (now unsupported) 1.2 works for many people, ask Steve.
So it's up to you.
Cheers,
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Grupo de Tecnología
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El 06/10/10 11:04, Zeeshan Zakaria escribió:
For a production environment, 1.4 is the most stable, and it has
everything
be unified to avoid time unit confusions.
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El 04/10/10 10:35, khalid touati escribió:
actually same thing happened to us a year ago (under asterisk 1.2) we
solved the same day discovered by putting both:
allowguest=no
alwaysauthreject = yes
Thanks for the tip!
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#aft_firmware
Regards,
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,
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http
recommend you to use Mozilla
Thunderbird...
Sorry for the offtopic.
And I agree, you should have no problems with asterisk using it inside
an openVZ VPS.
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as noted.
Go to your Avaya daddy...
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, but are you testing from a SIP
endpoint?
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] WARNING[3883]: loader.c:819 load_resource: Module
'func_aes.so' could not be loaded.
What are the requirements for these modules? Or is this an issue that
needs to be reported on the bugtracker?
Have a nice day.
Regards,
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Hi,
Never tried it, but you can take a look to the AUDIOHOOK_INHERIT
function that allows MixMonitor to continue the recording in the same
file after the transfer.
http://www.voip-info.org/wiki/view/Asterisk+func+AUDIOHOOK_INHERIT
Cheers,
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debugging,
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Hi,
Are you sure asterisk is receiving and processing DMTF correctly? Are
you using rfc2833, SIP INFO or inband DMTF? What is your asterisk
version? I use WaitExten(5) all the time, no matter if they are
single-digit or multiple-digit extensions.
Regards,
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Grupo de
El 09/08/10 05:30, michel freiha escribió:
Hello Miguel molina,
I did what you asked, but still the voice is too bad
Regards
On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina
mmol...@millenium.com.co mailto:mmol...@millenium.com.co wrote:
El 05/08/10 14:50, Tim Nelson escribió
on the iax.conf sample file...
disallow=lpc10; Icky sound quality... Mr. Roboto.
Cheers,
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of MELPe, anyone needing bitrates 2400 or less
should not be using LPC10.
-Jeff
OK, on years I have working with asterisk I never have used, tested or
even heard that old codec. I was just quoting the funny comment...
Cheers,
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pretty sure the Sangoma tech support will ask you you
upgrade the firmware to the latest version as a first measure.
Cheers,
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Grupo de Tecnología
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the
transfer outside the AGI, simply do a Goto after the AGI to transfer the
call where you need.
Even asterisk itself gives you help:
*CLI agi show set context
*CLI agi show set extension
*CLI agi show set priority
Cheers,
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to the kernel sources, just as I saw it
on a physical server. That way it worked compiling DAHDI.
I didn't know about explicitly setting the KSRC to the kernel sources.
Thanks for the hint.
Cheers,
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Grupo de Tecnología
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calls or
something directly into a VPS, or better, split channel groups between VPS.
Cheers,
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Grupo de Tecnología
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working after a while. Agents are able to
transfer or hangup a few calls and then it stops working. Doing some
debugging I could see that asterisk knows (receives) the DMTF too but
the features are not triggered...
Anyone else has run into this?
Regards,
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Grupo de
El 24/06/10 05:05, Tzafrir Cohen escribió:
On Wed, Jun 23, 2010 at 04:27:20PM -0500, Tilghman Lesher wrote:
On Wednesday 23 June 2010 15:45:05 Miguel Molina wrote:
Hi all,
Anyone know why this happens?
Mem:524288k total, 508120k used,16168k free,0k buffers
in caches) in 81858 allocations
Anoyone knows why the memory leak is not shown in the asterisk malloc
debug, and how can I figure what's causing it? The asterisk version is
1.6.2.9.
Thanks in advance.
Regards,
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Grupo de Tecnología
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El 21/06/10 14:04, Necati Demir escribió:
This is a really rookie question: when should i use TE110P ISDN PRI Card?
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---
When you need to...
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background noise at all. If
this is documented, point me to where and I'll gladly do my reading.
Thanks,
--Greg
Hi,
Look for rxgain and txgain config options in
http://www.voip-info.org/wiki/view/chan_dahdi.conf
Regards,
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', but the absence of mention seems
significant.
Take a look at http://sourceforge.net/projects/agx-ast-addons/
There is fax support for 1.4 inside that modules.
Cheers,
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PBX: (+57 1)6500800 ext. 1201
Fax: (+57 1)6500816
Móvil: (+57
have experience
with AGI but fastagi dont
This is the third thread you have created for this. You're boring me now.
S
Why don't to try it in Java? Use the asterisk-java library
(http://www.asterisk-java.org/) and you can run the FastAGI server in
any OS.
Cheers,
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Unreachable
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for it to
recheck the local channel member definitions.
Hope it helps.
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New
Gergo Csibra escribió:
Hello Asterisk,
This is only a test, because I can't start new thread in this list...
If you can send an email, you can start a new thread on this list.
What's the point of all this?
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Miguel Molina escribió:
Hi folks,
I am struggling to install cdr_pgsql in asterisk 1.6.0.26. When I do the
./configure, it complains about the function PQescapeStringConn not
existing in -lpq, so when I do a make menuconfig, I can't select the
cdr_pgsql module.
I am using CentOS 5.4
normally gets upset
with everyone that does this on the subject or in the body. I've
corrected the caps in the subject to avoid further upsetting.
Cheers,
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Grupo de Tecnología
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for 8.4 version. Some
of the installed packages are postgresql-libs, compat-postgresql-libs,
postgres, postgresql-contrib.
Any help would be very appreciated.
Thanks,
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Grupo de Tecnología
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PBX: (+57 1)6500800 ext. 1201
Fax: (+57 1)6500816
Móvil: (+57
, no matter if it's LTS on the new
release schema, will need time and a large user base that adopts it to
report bugs and help stabilize it. I would not underestimate the actual
1.6.X branches.
Just IMHO, any opinions welcome.
Cheers,
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Grupo de Tecnología
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);
00211 strcpy(amdStatus, HANGUP);
00212 break;
00213 }
So basically check that the channel is not being hungup during
application execution.
Regards,
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Grupo de Tecnología
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David @ULC escribió:
*Code:*
== Manager 'sendcron
it
is 10 digits in length.
My PRI provider needs it set to 10 digits always.
Thanks,
Jerry
*CLI core show function LEN
-= Info about function 'LEN' =-
[Syntax]
LEN(string)
[Synopsis]
Returns the length of the argument given
[Description]
Not available
Cheers,
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Neha Khandelwal escribió:
/_Our Product offerings:
_/
//Use the asterisk-biz list instead to advertise your asterisk-related
products.
Regards.
//
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would be changing this:
exten = _X.,1,Dial(SIP/${EXTEN})
To this:
exten = _X.,1,Dial(SIP/${FILTER(0123456789,${EXTEN})})
If you're intended to receive only numbers from the dialstring, right?
See http://www.voip-info.org/wiki/view/Asterisk+func+filter
Regards,
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Grupo de
,
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a maximum of 9600bps.
Cheers and Merry Christmas,
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://downloads.oreilly.com/books/9780596510480.pdf
The only thing is to configure the appropriate extensions to Dial()
through the DAHDI trunk between servers.
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the rest
of the configurations on this link:
http://www.voip-info.org/wiki/view/chan_dahdi.conf
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asterisk
.
Any idea what could be the possible reasons!
/ag
Please provide the asterisk version and g729 codec that is installed on
each server, so people can have a clue of what's happening. Maybe could
be a known bug or something.
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be
configured to do that?
I'm using asterisk 1.4.22 on a production server.
TIA,
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Hi Danny,
Seems reasonable, but I can't use that because I'm using permanently
logged in (AgentLogin) type agents, and that would require the use
Zeeshan Zakaria escribió:
Hi Danny,
This is exactly what I am doing, but it takes a few seconds before all
the extensions are ringing. The loop takes its time.
Are you originating the calls asynchronously?
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file solution Danny proposed would work pretty well in your case.
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Hi all,
Another simple question: does it make sense to use the append option in
MixMonitor (,a) when the codec is gsm? Or it works only when the codec
is an uncompressed one like ulaw, alaw or slin?
Thanks,
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with *,
it would end up in a blind transfer for the second agent who takes the
transferred call. Is there a feature that can be configured to do that?
I'm using asterisk 1.4.22 on a production server.
TIA,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
/asterisk-users
Hi,
Some months ago there was a discussion about this, with a simple
solution involving minimal changes to the source (1 line of code).
Search the archives of this list and you will find the answer. BTW, what
version of asterisk are you using?
Cheers,
--
Ing. Miguel Molina
Grupo
/manager_1_1.txt
Hope it solves your issue.
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it was.
That's all I can think of, without a hard work of messing with
app_queue.c source code. Hope you get the idea.
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language.
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,
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asterisk-users mailing
. Miguel Molina
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autopause = no
maxlen =
joinempty = no
leavewhenempty = no
reportholdtime = no
musicclass =
call-limit = 20
member = SIP/100
member = SIP/101
member = SIP/102
Please help , I m in a total mess ...Thanks Sriram
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-10 days...
Thanks for any pointers or help.
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Ekelund, Bryan escribió:
Upon further review, it is not dumping out, just restarting on its own with
the same error. No .dmp in /tmp
Check that you are running asterisk with the -g option.
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()
Or answer(1000).
Cool, didn't know about that one. One less line of code in the dialplan.
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Hi,
I didn't notice on my first answer, but we are on the -dev list and this
is not related to asterisk code developing. I will answer you on the
-users list, so we can continue the discussion there.
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Anahi Ludueña
to will tell
you if it's bridged or not, and to what channel.
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)
exten = s,n,Noop(I returned!)
exten = s,n,Hangup
[mysub]
exten = s,1,Noop(So I'm at a subroutine)
exten = s,n,Noop(I need to do special steps)
exten = s,n,Playback(tt-monkeys)
exten = s,n,Return()
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Anahi Ludueña escribió
is ${var1})
exten = 8135551212,n,Noop(var2 is ${var2})
exten = 8135551212,n,Noop(var3 is ${var3})
...
and so on... with no need to call Macro() or Gosub().
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Anahi Ludueña escribió:
Thanks,
I asked you to execute the GoSub from
for example.
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,
This should help: http://www.voip-info.org/wiki/view/Asterisk+log+queue_log
At the end you will find the meaning of every field.
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, googling error messages, dig into the bugtracker
looking for reported issues, and don't give up, is better to have or
achieve a stable version and maybe help to improve it reporting a new
bug than just going with a lazy solution IMHO.
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easy. When you see that asterisk works and that can do the recordings
and much more, you would start thinking on making asterisk your main
PBX
solution and leaving that legacy PBX for minimal uses.
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Erik de
Carlos Chavez escribió:
I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static
configuration for extensions.conf will not load.
Just curious, is there any specific reason for you to upgrade from the
latest 1.6.0.14 to 1.6.1?
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it to your needs.
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and that can do the recordings
and much more, you would start thinking on making asterisk your main PBX
solution and leaving that legacy PBX for minimal uses.
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version
into production.
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. RJ48C/RJ48C crossover cable specifications
That'll do.
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Hi,
Maybe maxlen = 1?
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and initial setup are far away difficult to understand (IMO). More than
that, you won't find anything else on the scope of Open Source dialers
for asterisk (AACC - Hanashi Dialer is in a very alpha stage). Anything
else is closed and/or commercial.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
that it supports.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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harry R escribió:
mysql -uasterisk -pasterisk asteriskdb
When I do that in a linux terminal it works.
But I always have this err 2002.
Greeting missing.
Elaborate missing.
Err 0x1b5a9f4c
You're not talking to machines here. :-)
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone
another files) to make it do both types of logging, text file
and realtime engine. A backend modular system similar to the CDR
handlers actually present in asterisk, would be awesome to handle the
queue logs too.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
Lenz Emilitri escribió:
You should log to a file and use a piece of code like our qloaderd to
do the DB update.
l.
Could you share such piece of code?
Thanks,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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the trick.
And PLEASE do not make more threads of this, is you are not satisfied
with the answers because of your lack of understanding, at least reply
on the same thread, giving more details about your setup and making an
effort to understand what people is trying to explain.
--
Ing. Miguel Molina
ISDN PRI to SIP or IAX2 gateway. Modify the
dialplan patterns according to your needs. For your PRI zaptel.conf and
zapata.conf there's plenty or info on the web to setup it.
Hope you get the idea.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
kind? Or are you checking
the plain CSV CDR file?
Supposing that you have a MySQL backend, you should have the userfield=1
setting in cdr_mysql.conf to tell the backend to save the userfield of
the record.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
the
advantages of web development, the old propietary PBX activity codes
seem to be obsolete nowadays.
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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= 999,1,Playback(tt-weasels|noanswer)
exten = 999,4,Hangup()
For incoming calls to 997 a CDR will be written, but not for 999.
How can I change this behavior?
Thanks
Klaus
Try unanswered = yes on cdr.conf
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
some module dependencies issue. What's the output of
ldd /usr/lib/asterisk/modules/res_speech_lumenvox.so?
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
___
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.
Regards,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
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