You would be much better off trying this question on the asterisk-users
list.
Much more traffic and geared towards Asterisk in general :)
-Brandon
John Cheng wrote:
Maybe I just haven't thought of the right google search terms -- but is
there a website/guide out there that will help me
Personally, instead of doing all that, I would simply add _one_
provider, then
when a call comes in on one DID, send it to one context/IVR, and when
the other
call comes in on the other DID, send it to it's own context/IVR.
I have done setups like this for bandwidth with 40+ DID's, you only
branches/2.0
All you have to do is click 'Add Service Provider'.
-bk
Klaus Ruebsam wrote:
Which is the correct GUI vor Asterisk 1.4 ?
Is it http://svn.digium.com/svn/asterisk-gui/branches/1.0 or
http://svn.digium.com/svn/asterisk-gui/branches/2.0 ?
Is there any kind of documentation for
Now this, even though not on the bug tracker, is a great example of a
bug report.
Because of that error code, I was able to easily fix the bug.
Fixed in revision 3671
-bk
Klaus Ruebsam wrote:
Hi list!
The topic-line already describes it all. I´m in the process of setting
up Asterisk
Works for me, try again.
-bk
Jerry Geis wrote:
This links: http://downloads.digium.com/pub/telephony/dahdi-linux-complete/
appear broken. thy just take me back to /pub
nothing downloads.
Jerry
___
-- Bandwidth and Colocation Provided by
That's pretty cool, I love ruby. What method does it use to
communicate to Asterisk? Does it manipulate raw config
files or use manager or something similar?
I am curious :)
-Brandon
Mike Clark wrote:
Our company, WebPoint IT Solutions has just released an open source (GPL
V2 license), Ruby
*Subject:* Re: [asterisk-users] Asterisk end-user GUI?
bkruse wrote:
I would checkout Switchvox :)
http://www.digium.com/en/products/switchvox/
-Brandon
-Ken
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update
Hey Guys,
Asterisk Trunk had some changes made about 7-8 months ago, that is also
in 1.6.x that
hindered the GUI from working (because of more strict config file
writing, which
is exactly what we needed to not fall into situations we had before).
This work was done for one main reason:
When
I would checkout Switchvox :)
http://www.digium.com/en/products/switchvox/
-Brandon
Ken D'Ambrosio wrote:
I badly want to roll out Asterisk at my job. Unfortunately, my boss is
dazzled by shiny objects. We had a vendor in today who showed us their
system which, honestly, didn't suck -- but
Eric ManxPower Wieling wrote:
Make the card stop sharing it's IRQ with your IDE controller. Try
moving the card to another slot.
Asterisk has to send an audio packet every 20ms for VoIP calls. I
believe Zaptel expects no more than a few ms of latency. If something
is causing a delay,
Doug Crompton wrote:
I saw that bug. Most of my files are WAV though. Would it apply to them
also?
Doug
On Tue, 1 Jul 2008, Noah Miller wrote:
Hi Doug -
In my research it appears this often happens when using more than one
processor. I am using a dual core Pentium.
I guess
I actually committed that patch to trunk/
-bk
Date: Sat, 28 Jun 2008 00:01:12 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Do not update to Firefox 3, yet?
On Fri, Jun 27, 2008 at 10:02:11AM -0400, Fidel Garcia wrote:
Yesterday I
Yes, probably, same basic error.
-brandon
Fidel Garcia wrote:
Great info! Thanks!
However, they do not mention the fact that when you create a new user you
cannot select the DialPlan. I wonder if the path fixes both issues. Any
idea?
Fidel Garcia
System Engineer
sysTeam.
7205 NW
to SSH (tcl/expect) and SSH.
Anyways, just my 2 cents,
-bk
Steve Totaro wrote:
bkruse wrote:
I am looking at Thirdlane's solution now. Very impressive and modest cost.
The asterisk GUI is free :]
I am not making any GUI purchasing or GPL decisions until I
(mass
deployment), i.e. SIP settings, programmable keys, ringtones, ...
All users are free to log in/out on their handsets or login at a
different handset and have their private phonebook etc. available
there (largely depends on the model).
bkruse wrote:
I have seen many setups
? Is that
customizable in the elusive html.conf file?
Any GUIs that are easily installed on existing systems and work with 1.2.x?
Thanks,
Steve
bkruse wrote:
svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow
thegui; cd thegui; sh configure; make sudo make install ; clear
I am looking at Thirdlane's solution now. Very impressive and modest cost.
The asterisk GUI is free :]
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
and modest cost.
Thanks,
Steve
bkruse wrote:
As Tzafrir stated, it will NOT work with 1.2.x.
Where is this html.conf, which README? I will update it.
I will write a brief page on setting up the *GUI for all who want to
know..
There are SOME GUI's that work with 1.2, however, I almost
svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow
thegui; cd thegui; sh configure; make sudo make install ; clear ;
echo 'completed'
-bk
Yann JOUANIN wrote:
You can do it from svn server , I think there is a page in the wiki
Best,
yann
Adhearsion is awesome.
-bk
Tzafrir Cohen wrote:
On Thu, Aug 16, 2007 at 08:30:43AM -0400, Mike Clark wrote:
Tzafrir Cohen wrote:
On Wed, Aug 15, 2007 at 06:12:09PM -0400, Mike Clark wrote:
Are there any nice GUIs out there for Asterisk Realtime? Google doesn't
yield
I have not had a chance to look at this, but if it is a fully
functional and threaded iax load balancer, thats extremely cool.
-bk
Stanisław Pitucha wrote:
Very low chances for that module if any. I haven't been using OpenSER much
and I don't think I'll be using it soon - but who knows.
This belongs in the asterisk-gui mailing list.
However, I will see what I can do.
-bkruse
FYI. It is just a javascript pattern matching function, its super easy
to change.
Tom Lobato wrote:
Hi all!
Is there a way to asterisk-gui to allow underline (as such cpd_tom)
in Names
Gotoiftime()
core show application gotoiftime
Thats the best bet it sounds like, but your question was
kind of hard to understand exactly, or why you would want to
do this.
-bkruse
Goke Aruna wrote:
Hello all,
I have a set up that answer my customer. and its working well,
however
dropped over 200 in at the same time (well, as fast
as mv can write them.) And the asterisk box went through all of them.
So if its a time thing, you should not have to worry about it.
-bkruse
shawn bright wrote:
hello there all,
if i have a script that writes drop files into
/var/spool/asterisk
Not sure, Talk to tech support and get the info.
You can always just go to another provider and transfer
your DID.
-bkruse
Todd H wrote:
I see that stanaphone is not accepting new customers. Does anyone
know if they are doing ok? I have a number with them and would like
to start
into graphs and make them look nice and what not.
If your interested in helping/doing this for me, email me at:
[EMAIL PROTECTED]
Thanks Guys!
so far this plugin is Rockin!
-bkruse
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
http://bugs.digium.com
please log it and bt. sounds fairly reproducable
Thomas Kenyon wrote:
On upgrading 2 machines (1 with a very simple configuration) from
asterisk 1.4.2 to 1.4.3, I have found that on receiving a call (on
either an IAX2 or SIP channel) the server process segfaults.
Is
Are your agents logged into the queue?
-brandon
Tim Verscheure wrote:
Hi,
I've been configuring AsteriskNOW from the GUI but could it be that
the GUI isn't working properly? because when I make a queue and add a
few agents, and when I call the queue none of the phones ring. The
queue is also
Sip Debug,
But I can tell you now that one of them is requesting g729, or, asterisk
has g729 set for one of its codecs in sip.conf and needs to translate it.
grep -r g729 /etc/asterisk/*
Alex Balashov wrote:
On Fri, 20 Apr 2007, Bruce Ferrell said something to this effect:
set_format:
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bkruse
Sent: Friday, April 20, 2007 15:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue problems
Are your agents logged into the queue?
-brandon
Tim Verscheure wrote:
Hi,
I've been configuring
ulimit
and ulimit -n
Matthew J. Roth wrote:
Patrick wrote:
[snip]
Just a wild guess because I don't really have an idea what is causing
this but are your ulimit settings high enough?
Patrick,
We have the maximum number of open file descriptors set to 65536. Are
there any other resources
${CALLERID(num)}
or
${CALLERID(name)}
Sanjay Rajdev wrote:
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find the same
in extensions.conf for setting a proper dialplan.
Please Suggest
Regards,
Sanjay Rajdev
___
--Bandwidth and
32 matches
Mail list logo
