none ?
2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com
Hi,
Have you noticed a marked increase in CPU load when using MixMonitor?
I use PHPAgi and Asterisk 1.6.2.9-2.
Mickael.
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On 5 Nov 2010, at 01:22, Mickael MONSIEUR wrote:
Have you noticed a marked increase in CPU load when using MixMonitor?
Since when? 1.6.2.9-1? 1.6.2.8? 1.0?
S
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Hi, one way to solve the problem with Mailbox or that Message that get's played
when busy/not available (same happens with Orange in Austria and other
providers) you can implement something similar to what Elastix/FreePBX has.
Confirm call - this will let the caller think it's still ringing
Dear Paul,
I submitted the issue to the tracker.
ID 0018263
Thanks
pepesz
On Thu, Nov 4, 2010 at 8:46 PM, Paul Belanger
paul.belan...@polybeacon.comwrote:
On Thu, Nov 4, 2010 at 3:24 PM, pepesz pep...@gmail.com wrote:
snip
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
El 04/11/10 17:14, Tzafrir Cohen escribió:
In the 'files' mode Asterisk plays the music separately for each
channel. If you use mpg123 or any other streamer, there is a single
stream per class.
A single stream per class sounds like good efficiency. Could you please
tell me what streamers can
On Fri, Nov 5, 2010 at 1:18 AM, Bruce B bruceb...@gmail.com wrote:
Chad,
You are absolutely right on this one. I had setup the Queue time out for
agent set to 15 seconds and retry to 2 seconds. So, I think during those two
seconds Asterisk for some crazy reason hits another extension and then
Hi,
marked - noticed.
I do not know where it comes from, my CPU goes from 2% to 60-70% at a
command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU
e4...@2.40ghz
2010/11/5 Norbert Zawodsky norb...@zawodsky.at
Am 05.11.2010 10:16, schrieb Mickael MONSIEUR:
none ?
This is just a brief reminder that today's VUC call will be about cloud
computing with some emphasis on voice applciations:
We have assembled a small panel of experienced people to discuss the
mattering, including:
Eric Chamberlain, Founder of RF.com,
Presenter to Astricon 2009 on running
Hello,
this is a test to add a channel to multiple GROUPs.
this is my dialplan :
exten = s,n,NoOp(groepcount = ${GROUP_COUNT(40)})
exten = s,n,Set(GROUP(40)=40)
exten = s,n,NoOp(This channel is member of : ${GROUP_LIST()})
exten = s,n,NoOp(groepcount = ${GROUP_COUNT(40)})
exten =
Yeah, I think I had it set to 2 seconds and that creates that short ring on
another extension.
Thanks,
On Fri, Nov 5, 2010 at 9:47 AM, Mark Deneen mden...@gmail.com wrote:
On Fri, Nov 5, 2010 at 1:18 AM, Bruce B bruceb...@gmail.com wrote:
Chad,
You are absolutely right on this one. I had
Friends,
After listening to Mark Summer's keynote at Astricon (hopefully soon on the
Astricon web site) I think we should come back to the discussion he started.
Mark talked about using Open Source in general and Asterisk in particular in
third world projects as well as in emergencies in other
Hi list,
My need is to append a site specific parameter to the
Contact: header on all INVITEs exiting * via a SIP trunk.
I'd like it to look something like this:
Contact: bob:3125551...@10.10.10.10;SITE-ID=us.here
where SITE-ID=us.here is set in a config file that * parses on
startup. Or in
Hi Gang,
My production box with my DAHDI cards is a 1.4.26 build. I
have 3 test machines that I do IAX communication with.
Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30.
Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1
VM running
Hi, please forgive me for this (hopefully) simple question. I cannot seem to
find an answer or solution while searching around.
I want to be able to call in to my server using my cell phone and be able to
set call forwarding for my extension and enter a phone number and also be
able to call in
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal
Sent: Friday, November 05, 2010 10:11 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Elementary question - accessing feature codes
fromcell phone
Hi,
I installed BigBlueButton and I want to change default conference playbacks. Is
it possible ?
if Yes, how :)
Thank you.
Bu elektronik postada bulunan tum fikir ve gorusler ve ekindeki dosyalar sadece
adres sahip/sahiplerine ait olup, Yasar Toplulugu Sirketleri bu mesajin icerigi
ile
Hi,
We want to upgrade both our servers to asterisk 1.8, the one from Romania and
the one from Chicago, but for the moment I`m trying to install Asterisk 1.8 on
a test machine running CentOS 5.5 with the kernel: Linux asterisk3
2.6.18-194.17.4.el5PAE #1 SMP Mon Oct 25 16:35:27 EDT 2010 i686
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Erol Demir
Sent: Friday, November 05, 2010 10:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk default sound files
Hi,
I installed
On Fri, Nov 5, 2010 at 10:38 AM, Bruce B bruceb...@gmail.com wrote:
Yeah, I think I had it set to 2 seconds and that creates that short ring on
another extension.
Thanks,
The point was that 14 and 16 are divisible by 2 (evenly) while 15 is not.
--
On Fri, Nov 5, 2010 at 11:04 AM, Danny Nicholas da...@debsinc.com wrote:
Hi Gang,
My production box with my DAHDI cards is a 1.4.26 build. I
have 3 test machines that I do IAX communication with.
Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30.
Machine 2
DTMF sent from cell phones are usually not well recognized at the asterisk
end. The main reason for this is that cell phones transmit out-of-band DTMF,
which by the time reaches an asterisk server traveling through cell towers,
their equipment, various VoIP carriers etc. is usually drifted away
It worked! I['ll have to figure out how to add the dial string to the
phone.
Thanks a bunch for your help
On Thu, Nov 4, 2010 at 9:04 PM, Mark Phillips g7...@g7ltt.com wrote:
I would second that.
If you don't set a dial string in your handset then it waits for N
seconds before submitting
Thanks for the quick response! I have had a lot of issues in the past with
DTMF.
Anyway, I think the idea of replicating the function into an extension will
work. Any pointers on the best way to accomplish this? I created a new
extension but am unsure what to do next. I thought about the FollowMe
Hi all,
I'm trying to establish jingle call in this network scenario:
Asterisk - NAT - Internet - HTTP_PROXY - GTalk client
The call is received and answered in gtalk but there is no audio in the call.
I suppose it could be related to the support for relay candidates in asterisk
jingle
On Fri, Nov 05, 2010 at 12:49:45PM -0400, John Regal wrote:
Anyway, I think the idea of replicating the function into an extension will
work. Any pointers on the best way to accomplish this? I created a new
extension but am unsure what to do next. I thought about the FollowMe
feature but I would
Thanks for the answer.
All of those libraries are already installed and it's still not working.
Package libstdc++-devel-4.1.2-48.el5.i386 already installed and latest
version
Package matching libxml2-devel-2.6.26-2.1.2.8.i386 already installed.
Checking for update.
Package
I`ve disabled chan_ooh323 and res_adsi and it worked .
Bogdan
-Original Message-
From: Bogdan Sarandan
Sent: Friday, November 05, 2010 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8 Installation Problem
Thanks for the
Sorry, I am not following. If an extension rings for 15 or 16 seconds and
then waits for 2 or three seconds what difference does the being divisible
make?
Is there something internal to Asterisk that makes the Retry time dependent
on Time Out (also known as Ring Time)?
P.S. I think the 15
We use DISA (http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA) to access
our entire [features] context from our cell phones.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal
Sent: Friday, November 05, 2010 11:11 AM
To:
Hi Everyone,
Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For
some reason I don't see any SIP packets coming in to Asterisk at all. I
don't want to use XML or ftp etc for now and just use the Web Interface to
get it going with basic features. But the Web UI is a bit
Hello,
I'm having a problem trying to do this, and it used to work with Asterisk
1.4.
Since Asterisk 1.6 series I have not been able to place more than one call
to a DID.
I get this message:
Skipping dialing interface 'SIP/16034817...@flowroute' again since it has
already been dialed
I'm
Hello,
I'm having a problem trying to do this, and it used to work with Asterisk
1.4.
Since Asterisk 1.6 series I have not been able to place more than one call
to a DID.
I get this message:
Skipping dialing interface 'SIP/16034817...@flowroute' again since it has
already been dialed
I'm
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Frager
Sent: Friday, November 05, 2010 2:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Unable to place 2 or more calls to a DID
Hello,
I'm having
Hi Everyone,
Is there other comparable products to Proxmox to be used for Asterisk
instances? Ease of use, web interface, and Asterisk/CentOS support would be
ideal.
Thanks
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- Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
Is there other comparable products to Proxmox to be used for Asterisk
instances? Ease of use, web interface, and Asterisk/CentOS support would be
ideal.
There is OpenNode:
http://opennode.activesys.org/
I've heard good things
- Tim Nelson tnel...@rockbochs.com wrote:
Hi Everyone,
Is there other comparable products to Proxmox to be used for Asterisk
instances? Ease of use, web interface, and Asterisk/CentOS support would be
ideal.
There is OpenNode:
http://opennode.activesys.org/
I've heard good
Hi,
I'm trying distributed events with Openais but don't work.
I made the test with two asterisk box in the same LAN
box A: 192.168.142.246 asterisk 1.6.2.13
BoxB: 192.168.142.248 asterisk 1.8.0
openais.conf:
# Please read the openais.conf.5 manual page
totem {
version: 2
secauth: off
On Fri, Nov 5, 2010 at 4:44 PM, bakko asannu...@gmail.com wrote:
When I load res_ais.so module, the pbx crash (boths)
Generate a backtrace[1] and upload to this thread.
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
On Fri, Nov 5, 2010 at 10:34 AM, Mark Deneen mden...@gmail.com wrote:
On Fri, Nov 5, 2010 at 11:04 AM, Danny Nicholas da...@debsinc.com wrote:
Hi Gang,
My production box with my DAHDI cards is a 1.4.26 build. I
have 3 test machines that I do IAX communication with.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Friday, November 05, 2010 4:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Funky IAX behavior between 1.4 and
Hey, all. I'm in the middle of a rollout, and just learned that the
SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued.
The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130
more/handset. AND it doesn't look as nice.
Ouch.
Does anyone have any
On Nov 5, 2010, at 5:35 PM, Ken D'Ambrosio wrote:
Hey, all. I'm in the middle of a rollout, and just learned that the
SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued.
The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130
more/handset. AND it
Hi,
after some test the system don't crash but no members:
CLI ais show clm members
=
=== Cluster Members =
=
===
===
Hi,
What is the easier way to make call using a password? I have A2billing but
its authentication is too big, I would like four digits long. Something like
that: In any extensons, the user dial the password and make call.
Thanks in advanced!
Att,
Flavio Roberto Miranda
On Nov 5, 2010, at 10:45 PM, Michael Graves wrote:
On Fri, 5 Nov 2010 19:02:43 -0400, Mike wrote:
On Nov 5, 2010, at 5:35 PM, Ken D'Ambrosio wrote:
Hey, all. I'm in the middle of a rollout, and just learned that the
SoundPoint IP 430 -- my favorite mid-range phone -- has been
On Fri, 5 Nov 2010 23:09:19 -0400, Fred Posner wrote:
Curious Michael... Why won't you subject people to the 335's? I love these
phones for a call center deployment. The are a fantastic agent phone... let
alone a great phone for kitchens, break rooms, lobby, etc. But I love them as
an agent
On Nov 5, 2010, at 11:13 PM, Michael Graves wrote:
On Fri, 5 Nov 2010 23:09:19 -0400, Fred Posner wrote:
Curious Michael... Why won't you subject people to the 335's? I love these
phones for a call center deployment. The are a fantastic agent phone... let
alone a great phone for kitchens,
The subject says it all. I'm betting there's a way to do it, but so far
I haven't found the dialplan runestone via web searching.
Thanks.
b.
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Hi
I am trying to get the recall button working for the gigasets
What settings do i need to set in the advance settings?
Zakir--
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