Dean Collins wrote:
ADA Forums: http://forums.digium.com/index.php?c=8
ADA Download: http://dl1.digium.com/ADA/ADAInstall.exe
ADA Administrators Guide: http://dl1.digium.com/ADA1.1/ADA_Admin_Manual.pdf
Thanks for the links. I hadn't seen that before. The product is kind
of interesting, but
2009/3/6 BERGANZ François franc...@acropolistelecom.net
hello,
I will do a server to do a lots of conferences (MeetMe).
I want to know that if I dont use a digum card, the limit of simultaneous
calls is harder without a card than with a card ?if, yes, how harder is the
limit?
Hi Kevin,
Kevin P. Fleming kpflem...@digium.com hat am 5. M�rz 2009 um 14:10
geschrieben:
Has anybody set up such an installation and/or is OpenAIS able to
transfer the devstates over different subnets? Haven't found docs and
hints for this use case.
The method OpenAIS uses to
Watkins, Bradley bradley.watk...@compuware.com hat am 5. M�rz 2009 um 16:46
geschrieben:
The method OpenAIS uses to communicate between nodes is
designed for a
very low latency local connection; it is not designed to work across
routed connections. Russell Bryant has spent some time
Hi,
Any ideas why? If I leave it out - there is ring tone passed through.
Using g729 codec. Sip based call...___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hello !
I've upgraded our testsystem from asterisk 1.4.21 to asterisk 1.6.0.6.
We 've noticed that the log files are now in colour.
I could not find a note in the upgrade section about this.
Is this a feature or a bug ?
It might be usefull to have them not in colour.
best regards
Hans
On 6 Mar 2009, at 08:58, Shaun Wingrin wrote:
Hi,
Any ideas why? If I leave it out - there is ring tone passed through.
Using g729 codec. Sip based call...
What version of asterisk, what licenses, what endpoints, what
transcoding?
S
___
--
Well, i can share mine backports of queue_log into mysql for 1.4.
Basically you need two backports (that's why there are numerous
files). Realtime store/destroy allows Asterisk Realtime engine to use
INSERT's on MySQL. It needs two patches - one for Asterisk, one for
Asterisk-addons (mysql part).
I just installed 1.4.23.1 with the queue realtime logger backport. Here
are my configs:
musiconhold.conf
[default]
mode=files
directory=/var/lib/asterisk/moh-native
random=yes
queues.conf
[7703]
wrapuptime=0
timeout=15
strategy=rrmemory
retry=5
queue-youarenext=queue-youarenext
Great backports! :-)
This should really be merged into 1.4.
--
Regards,
Robert Broyles
Atis Lezdins wrote:
Well, i can share mine backports of queue_log into mysql for 1.4.
Basically you need two backports (that's why there are numerous
files). Realtime store/destroy allows Asterisk
Hi all,
Is there any way to make use of the SIP making progress messages?
I find that about the time the SIP peer says making progress is the
time the other end actually starts to ring, or is busy etc.
Before that time, I want to generate an in progress tone using
playtones to let the user know
the transcoding card isnt a good source for timing. the card only make
interruptions if it is working.
if the meetme dont requeire transcoding the card will not generate any
timing.
David
2009/3/6 Grygoriy Dobrovolskyy megaho...@gmail.com
2009/3/6 BERGANZ François
Occasionally, DIDs from different providers stop working for some reason.
I would like to be able to monitor situations like that and react before any
of my clients start going ballistic on me.
Any ideas? Scripts you know of or wrote and willing to share?
Any info would be greatly appreciated.
I
Santiago Gimeno schrieb:
Hello,
Thanks for the reply.
Yes, I'm using pedantic=yes. I will report this asap.
One more thing that I have observed and might be also related to this issue.
The scenario is the same as the one I described in the previous mail,
but in this case, the SIP
I'm having trouble with call pickups.
Suppose ring group is 100 and has extensions 101 and 102.
Someone calls 100, 101 rings and 102 wants to pick the call up. If 102 dials
**100, call pickup works. If 102 dials **101, call pickup fails.
In my dialplan I have:
exten = **101,1,NoOp(pickup
Hi!
I have the following weird problem:
phones A,B and C are in the same callgroup/pickupgroup.
A call B, B is ringing, C calls *8.
Now, B is CANCELed, C gets 200 OK, but A is still in Ringing.
Is there anything else I have to configure?
thanks
Klaus
The log files themselves are not in color. It would be a style sheet change
on the GUI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johann
Steinwendtner
Sent: Friday, March 06, 2009 2:59 AM
To: Asterisk
--- On Fri, 3/6/09, Vieri rentor...@yahoo.com wrote:
I'm having trouble with call pickups.
Suppose ring group is 100 and has extensions 101 and 102.
Someone calls 100, 101 rings and 102 wants to pick the call
up. If 102 dials **100, call pickup works. If 102 dials
**101, call pickup
On Fri, 06 Mar 2009 02:15:12 +0800, Steve Underwood wrote:
Steve Underwood wrote:
They might be doing some kind of fake bandwidth expansion. You can't
create something out of nothing, and make the narrowband voice more
intelligible, but you can make it sound pretty nice. Jean-Marc Valin (he
Go to http://www.voip-info.org/wiki-Asterisk+tips+and+tricks and try some of
the Dial Plan solutions. You can probably find something to your liking
that will work with little or no tweaking.
_
From: asterisk-users-boun...@lists.digium.com
On Thu, 5 Mar 2009, Robert Augustyn wrote:
Occasionally, DIDs from different providers stop working for some
reason.
I would like to be able to monitor situations like that and react before
any of my clients start going ballistic on me.
Are you losing DIDs that terminate on your Asterisk
Just found out my maintanence agreement with fonality does not cover
phones over a year old. So, of course, they don't repaird phones, only
replace them. Very worldly of them...
Anyone know where I can get an Aastra 480i repaired? The phone works on
speakerphone, but when you lift the
You have KkTt on your Dial command?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion
Sent: Friday, March 06, 2009 8:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Klaus Darilion wrote:
Hi!
I have the following weird problem:
phones A,B and C are in the same callgroup/pickupgroup.
A call B, B is ringing, C calls *8.
Now, B is CANCELed, C gets 200 OK, but A is still in Ringing.
Is there anything else I have to configure?
thanks
Klaus
On Fri, 2009-03-06 at 09:41 -0500, m...@njycamps.org wrote:
Anyone know where I can get an Aastra 480i repaired? The phone works
on speakerphone, but when you lift the receiver offthe hook, the phone
does not engage. There is something wrong with the hook. The
receiver works fine, on another
Danny Nicholas wrote:
The log files themselves are not in color. It would be a style sheet change
on the GUI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johann
Steinwendtner
Sent: Friday, March
This is a feature. It seems to be the same in 1.4.21, 1.4.22 and 1.6, but
you could just change lines 150-155 in logger.c and recompile.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johann
Steinwendtner
Mark Michelson schrieb:
Klaus Darilion wrote:
Hi!
I have the following weird problem:
phones A,B and C are in the same callgroup/pickupgroup.
A call B, B is ringing, C calls *8.
Now, B is CANCELed, C gets 200 OK, but A is still in Ringing.
Is there anything else I have to configure?
Hi!
What are the typical ways to work around the 64 groups limit?
thanks
klaus
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Actually, I have a bunch of the 480i's and like them much better than any of
the Polycom phones I have. The screen is MUCH easier to read, the
phones, while def not as solid, are much easier to navigate for my end-users
and make changes to. IMO, of course... But thanks LOADS for that info. $25
On Fri, 6 Mar 2009, Klaus Darilion wrote:
Updating to 1.4 branch solved the issue. Thanks.
Pity that they still didn't release a new version that works properly.
1.6.0.6 is broken too, SIP doesn't work on 2 difference boxes i tried it
on.
___
--
Remco Barendse wrote:
On Fri, 6 Mar 2009, Klaus Darilion wrote:
Updating to 1.4 branch solved the issue. Thanks.
Pity that they still didn't release a new version that works properly.
We can't afford to release a new version every time we fix a bug. That's just
not practical.
1.6.0.6 is
Remco Barendse schrieb:
1.6.0.6 is broken too, SIP doesn't work on 2 difference boxes i tried it
on.
Let's continue that discussion about SIP in 1.6.0.6 in the
appropriate thread:
http://lists.digium.com/pipermail/asterisk-users/2009-March/228044.html
Thanks,
Philipp Kempgen
--
AMOOCON
Hi,
Does anyone have some good examples of a Kamalio or OpenSips
configuration that integrates with Asterisk?
Essentially I want to use the SIP router as the UA, but still run all
the calls through Asterisk (for dialplan, etc..)
I've looked for examples on the project web sites, but I haven't
On Fri, Mar 6, 2009 at 3:03 PM, Klaus Darilion klaus.mailingli...@pernau.at
wrote:
Santiago Gimeno schrieb:
Hello,
Thanks for the reply.
Yes, I'm using pedantic=yes. I will report this asap.
One more thing that I have observed and might be also related to this
issue.
The
I have a caller screen queue setup. Basically a caller calls in, goes
through a IVR, and before that caller is put into the queue, they get a sub
ran on them first asking for them to say there name. That gets saved and
they are entered into the queue using Queue(mainqueue300).
In the
Shaun R. wrote:
I have a caller screen queue setup. Basically a caller calls in, goes
through a IVR, and before that caller is put into the queue, they get a sub
ran on them first asking for them to say there name. That gets saved and
they are entered into the queue using
Mark,
If you set the GoSub in the queue() instead of dial it works, the problem i
had there was that GOSUB_RESULT didnt look to be looked at or listened to.
If i remember correctly i had to add the c option to queue() so that when a
member hung up with out selecting a option that it wouldnt
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins arob...@pharmacentra.comwrote:
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade.
On the server that SENDS the call, I have the following in SIP.CONF:
[192.168.1.2_OB]
no
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tracinet
Sent: Friday, March 06, 2009 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
This may be of interest -- as a tool we can use to test our systems and as
a weapon that may be used against us :)
http://warvox.org/
A brief read-over looks like it uses iaxclient and ruby to war dial a
range of numbers and record audio samples to be analyzed to identify if
the call
That stinks... We are migrating to SIP from IAX2 at the moment and running
into the same exact problem. No way to control the destination context
unless you use the fromuser. Of course that is rendering Caller ID
useless as you pointed out.
I am still researching this though, if I find anything
Hi ,
Can anyone suggest me how to start Early Media in asterisk .
--
Thanks Regards,
Rajkiran Reddy,
09825698439 ,
Ahmedabad , India .
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
I know the call parking feature changed in 1.4.23.1 to fix some serious
issues. I'm seeing a major change though which I find disturbing.
A person parks a call by transferring it to the parking position (700).
When the timeout value is reached, the call is NOT returned to that
device, but
On 6 Mar 2009, at 19:29, tracinet wrote:
That stinks... We are migrating to SIP from IAX2 at the moment and
running into the same exact problem. No way to control the
destination context unless you use the fromuser. Of course that
is rendering Caller ID useless as you pointed out.
Last thing we need is more war.
--
Sent from mobile device
On Mar 6, 2009, at 2:29 PM, Steve Edwards asterisk@sedwards.com
wrote:
This may be of interest -- as a tool we can use to test our systems
and as
a weapon that may be used against us :)
http://warvox.org/
A brief
Hi all, I’ve read that meetme works at G711 (ulaw), so asterisk would
down-mix a number of parties using G722, is that still correct?
If so, i’ve also read that Joshua Colp was/is working on a replacement
(conf_bridge?) that works with G722. If this is this still in active
development are there
Razza wrote:
If so, i’ve also read that Joshua Colp was/is working on a replacement
(conf_bridge?) that works with G722. If this is this still in active
development are there any planned timelines? If it’s in 1.6.0.6, and
i’ve just missed it or it’s been renamed please be nice in your
Just a silly question that I'm not sure.
Ringinuse is working with IAX in 1.6??? like in sip??
Thanks!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
It utilizes the iaxclient for piping the raw audio to a flat file where it's
then analyzed.
On Fri, Mar 6, 2009 at 11:43 AM, Alex Balashov abalas...@evaristesys.comwrote:
Last thing we need is more war.
--
Sent from mobile device
On Mar 6, 2009, at 2:29 PM, Steve Edwards
On Tue, Feb 3, 2009 at 2:58 PM, Jose P. Espinal j...@slackware-es.com wrote:
Hello List,
I have been working on a little PHP software that uses AMI's
UpdateConfig command in order to modify some of it's config files.
I was working with 'Asterisk 1.4.22.1' and everything was working.
After
Another war dialer with IAX capabilities:
http://www.softwink.com/iwar/
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Steve Edwards asterisk@sedwards.com wrote:
This may be of interest -- as a tool we can use to test our systems
and as
a weapon that may be
Just a suggestion: have you tried more recent versions of Asterisk
with IAX2? I'm uncertain what version you're using, and if it's
1.2.4, that's getting to be quite old and the problems that you
reference may already be solved in more recent updates.
In addition, if you're set on SIP,
Tim Nelson wrote:
The fact that this would be even being discussed on this list is an
embarrassment to the asterisk community.
I am constantly being pestered by cold callers with fake caller ids,
probe calls such as this, etc. I think for once CRTC/FCC need to step up
to the plate and take
Klaus Darilion klaus.mailingli...@pernau.at writes:
What are the typical ways to work around the 64 groups limit?
a) Split into different Asterisks
b) Use directed pickup instead, not *8
/Benny
___
-- Bandwidth and Colocation Provided by
Sebastian wrote:
Just a silly question that I’m not sure.
Ringinuse is working with IAX in 1.6??? like in sip??
I assume you're referring to the queues.conf option, correct? An easy way to
check is to issue a queue show command when an IAX2 queue member receives a
call. If his status is
On Fri, Mar 06, 2009 at 04:24:42PM -0500, Jon Pounder wrote:
Tim Nelson wrote:
The fact that this would be even being discussed on this list is an
embarrassment to the asterisk community.
Why? I didn't know about those dialers before.
This type of software is something that someone will
Ugh! That's a terrible feature, if that's unselectable. However, I
don't see that being the case in SVN-TRUNK as of a few days ago (well,
Feb 17th.)
My logfiles don't have any escape codes/color codes in them.
What version, exactly, are you using?
JT
On Mar 6, 2009, at 7:54 AM, Danny
One opcion..!!
cat /var/log/asterisk/messages |grep '#callerid=Helius' --color=auto
On Fri 06 Mar 2009 12:10:32 Danny Nicholas wrote:
The log files themselves are not in color. It would be a style sheet
change on the GUI.
-Original Message-
From:
Not to burst your bubble, Jon, as I agree with a majority of what you
said... but using an argument about the evolution of email to support an
argument about how telcos should have better tracking and accountability
is somewhat weird.
We get 3 million email messages a day through our servers.
SIP wrote:
Not to burst your bubble, Jon, as I agree with a majority of what you
said... but using an argument about the evolution of email to support an
argument about how telcos should have better tracking and accountability
is somewhat weird.
We get 3 million email messages a day
On Fri, Mar 06, 2009 at 09:59:18AM +0100, Johann Steinwendtner wrote:
Hello !
I've upgraded our testsystem from asterisk 1.4.21 to asterisk 1.6.0.6.
We 've noticed that the log files are now in colour.
I could not find a note in the upgrade section about this.
Is this a feature or a bug ?
Umm... Caller ID spoofing and DSP audio processing of called numbers are two
entirely different subjects.
And as far as creating more laws:
I say fix the damn technology first (Caller ID) before wasting tax payers
money on more laws on the books that will be obsolete in a few years time.
While
Basically, Server 1 is the main customer PBX where we have multiple
customers running (each on their own virtual PBX separated by their
contexts). Each customer has their own accountcode that we use to track
calls for billing purposes, etc. The customer uses a SIP phone to register
to Server 1
I am on Asterisk 1.4.23.1. What you propose is interesting. I will look
into this ASAP to see if this will help. Thanks!
On Fri, Mar 6, 2009 at 4:23 PM, John Todd jt...@digium.com wrote:
Just a suggestion: have you tried more recent versions of Asterisk
with IAX2? I'm uncertain what
On Fri, Mar 6, 2009 at 10:39 AM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
Danny Nicholas wrote:
The log files themselves are not in color. It would be a style sheet change
on the GUI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
In Canada the do not call registry is useless since calls do not
originate in Canada nor do the violators care if they are doing
something illegal, Telcos could take this further and if a number of
complaints are received about a call source, offer an opt-in blocking
plan to throw those calls
- Jon Pounder j...@inline.net wrote:
Tim Nelson wrote:
Now you're making others think I wrote this tirade of crap.
The fact that this would be even being discussed on this list is an
embarrassment to the asterisk community.
Isn't this the *perfect* place to be discussing this type
What version of ubuntu are you running?? That makes a difference...
But you need to install libc-client. On my system it is libc-client2007b and
libc-client2007b-dev
Once you install those packages, then do './configure --with-imap' and you
should be good to go...
Thanks,
Jeff Phelps
IT
the function SIP_HEADER and application SIPAddHeader seems to work
nicely upon initial testing... Thanks for the tip! Out of curiosity, are
they any standards in additional header names for the caller ID values I am
trying to add to the headers? I am using X-CID for now...
Thanks again!
On
Mis saludos a todos, estoy presentando un problema, tengo conectado un asterisk
con un IAD208 huawei, pero no logro la comunicacion de los telefonos del huawei
con el asterisk... asterisk me ve los terminales mgcp pero me da los siguientes
errores:
mi e-mail es obarr...@estudiantes.uci.cu
hi,
I'm working with asterisk on a project and I found a problem with cdr_odbc.
As we know, after answering each call a cdr event is raised which is saved
in cdr_csv and cdr_odbc. but here my point is on cdr_odbc. some information,
including start_time and end_time is given by cdr event but the
72 matches
Mail list logo
