Redhat Enterprise
Zeeshan Zakaria wrote:
I've to setup Asterisk on a Dell PowerEdge 1750 server. Its dual Xeon
3GHz with Hyperthreading. People on this list who have experience with
this server please advise me how is the performance of Asterisk on
this server, what flavour of linux is good
Hi
I have 100XP Digium clone card
Installed in my pc
and compiled zaptel and asterisk again after installing the card
but after i rebooted
i can load zaptel and wcfxo modprobe with out any problem
but when i intiated ztcfg -
i get the following error
Zaptel Configuration
Hi,
i have Asterisk 1.2.7.1 and outgoing trunk connected via SIP (this is Cisco
AS5350)and user is connected via sip too.
When user calling out (via AS5350) he receives progress tone generated by
voip-phone not that passing from telco line.
I turned on debug and see that the AS send: 183
On 5/26/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Thanks Shanon and everyones input...
Finally, got the application working as planned with PHPAGI...
Now the only draw back is the voice... I am using text2wav to prompt all
the questions, but the voice is creepy...
Is their any easier way
I am using asterisk 1.4.4 now and facing a problem with meetme,the code I
was using with asterisk 1.2 is not functioning with 1.4 ,my code is
conf = 222| at meetme.conf
at meet_me_additional
like this
exten = 21,1,MeetMe(21,dq)
exten = 21,2,Playback(beep)
or this
exten =
Hi,
On Sun, 2007-05-27 at 17:35 -0400, Nabeel Jafferali wrote:
Looks like a rebadged Patton 6075 to me:
http://www.patton.com/products/pe_products.asp?category=337
also patton rebrands that unit.
At Cebit there was plenty of these boxes from .tw manufacturers.
Matteo
--
Matteo Brancaleoni
Hello *,
do queues allow me to set an announce like the A() option of the Dial() cmd?
The announce that I've found is a message that is heard by the caller. I'd like
to send a message to the member of the queue that picks up the call.
Thanks in advance,
--
Dott. Andrea Spadaccini
Multimedia
hi list,
im setting up a realtime , and while entering datas in my extensions_table
one question came in my mind
how should insert the different includes ligns (includes=context)in my
database knowing that
im not going to use the extensions.conf file anymore
should i add thoses lines in the
Hi All,
I need to limit outgoing calls in my sip peers...
I tried to use call-limit=1 in these peers in the sip.conf, but it
didn't work...
Here is my peer configuration in the sip.conf:
[sip.broadvoice.com]
accountcode=broadvoice
type=peer
dynamic=yes
username=MYUSERNAME
fromuser=MYUSERNAME
What does codec has to do...? I am using G729a
Cheers,
Nitesh
ram wrote:
On 5/26/07, *Nitesh Divecha* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Thanks Shanon and everyones input...
Finally, got the application working as planned with PHPAGI...
Now the only draw
On May 24, 2007, at 3:28 PM, Doug Lytle wrote:
Paul Aviles wrote:
Hello guys,
I have been looking for a way to call a cell phone after someone
has left a
This can easily be done with database lookups and .call files
to accomplishing this? Most analog pbx's have this feature and I
am
Good morningI was wondering whether you could help me. I
installed sipp on my Asterisk server but I don't really understand how
does it fonction! Has someone ever tried it?If you can explain to me the
principle, I would be extremely grateful.Thank you very much in advance.
Hello!
If ou mean SIPp, the testing tool for the SIP protocol or kind of a call
generator for Asterisk PBX, have a look at
http://sipp.sourceforge.net/doc/reference.html
cheers
- Original Message -
From: khawla khawla
To: asterisk-users@lists.digium.com
Sent: Monday, May 28,
On Fri, 2007-05-25 at 17:17 -0500, JR Richardson wrote:
Hi All,
Call comes into Asterisk
Asterisk answers and Dials SIP Phone
SIP phone has call forward enabled to a long distance number
Asterisk receives a SIP response 302 Moved Temporarily back from phone
Asterisk then forwards inbound
Hello,
We are proudly to present new version of our billing system MOR v0.4
What's new in MOR FREE v0.4
* Extended stability and reliability
* Extended configuration options for clients and providers
* User blocking
* Prepaid support
* Increased security
* New
Hi,
I was looking for a way to pass alphanumeric variables to asterisk via
the keypad, found this application app_dtmftotext.c , And I already tried
with 'spandsp' application for this. But I am getting errors.[I followed the
instructions at http://www.soft-switch.org/installing-spandsp.html].
I've fired a script from an AGI-BIN to accomplish that.
Try this one:
#!/usr/bin/perl
# mk 2004 feel free to distribute
# [EMAIL PROTECTED], _Vile
# perl script to reboot phones
# try telnetting to your phone, first.
#
use Net::Telnet ();
$phone_ip = shift;
# Your Cisco 79xx prompt
How much does a Patton NanoServ 607x cost? Their page has no price, an
inactive Ordering tab, Google doesn't have (nanoserv 6070 price) in
its index (except a couple unresponsive del.ic.ious pages). PingTel
announce a SIPxNano based on it, for under $1000 in 2006Q3:
25 maj 2007 kl. 06.40 skrev JK:
Hello asterisk-users list.
I have been scratching my head for almost a week. We are trying to
set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not
working.
In our scenario the SP is sending call to our ser server and ser
is forwarding the
Hello,
We are creating a web-based conferencing application using Asterisk as the
voice conferencing server.
This as an open source project. We are trying to determine if there
is interest of the community and perhaps work together to improve the
application.
Using the web application, you can
I think this is a great potential application for Asterisk - I couldn't
actually determine if/where you had a downloadable POC or if it was
still just in development conceptualization at the moment.
Either way keep up the good work and put a paypal tip jar up once you
have something people can
Yes, we have some downloadable code. We are in the process of completing the
instructions (build/deploy/etc.).
Code is located here.
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk
Partial docs is located here.
Anybody??
-Original Message-
From: shadowym [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 24, 2007 9:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Bottom line on fax reception
So what is the bottom line? Does it work or not. I've heard stories it
works, it
At 17.09 28/05/2007, Richard Alam wrote:
Yes, we have some downloadable code. We are in the process of completing the
instructions (build/deploy/etc.).
Code is located here.
Sounds cool. You could probably use some code from the various open
source jabber clients that allow for shared whiteboard and pushing URLs
too.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Alam
Someone already answered this question. The answer is no, it does not
work by your definition of production ready.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of shadowym
change conf = 222
to conf = 222
( remove | )
I had same problem as freepbx always put | removing it fixed the problem
On 29/05/07, Khaled Chehab [EMAIL PROTECTED] wrote:
I am using asterisk 1.4.4 now and facing a problem with meetme,the code I
was using with asterisk 1.2 is not functioning
Hello all,
Some of you are using astmanproxy with asttapi or activa TSP?
How does you make to work?
Thanks
VoipCrazy
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi all,
Years ago, I was pretty sure attempting to use two TDM400p cards in one
machine was recommended against by Digium ... probably because the cards
couldn't hack it, and/or interrupt problems etc
I have seen some posts recently that seem to indicate it is in fact possible
these days thanks
On 5/28/07, Roberto Fichera [EMAIL PROTECTED] wrote:
At 17.09 28/05/2007, Richard Alam wrote:
Yes, we have some downloadable code. We are in the process of completing
the instructions (build/deploy/etc.).
Code is located here.
Hi Steve,
Yes, we are looking for that. Do you know of any projects that provides
those? I know one written in TCL/TK.
Thanks.
Richard
On 5/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Sounds cool. You could probably use some code from the various open
source jabber clients that allow
Hi,
I'm currently testing SoftEcho, an echo cancellation software for
Asterisk from Octasic. I noticed an important increase of the quality of
my coms, but I still have a few echo problems.
There is an ERL parameter which corresponds to an initial ERL value
probably to optimize the echo
At 19.19 28/05/2007, you wrote:
On 5/28/07, Roberto Fichera mailto:[EMAIL PROTECTED][EMAIL PROTECTED]
wrote:
At 17.09 28/05/2007, Richard Alam wrote:
Yes, we have some downloadable code. We are in the process of completing the
instructions (build/deploy/etc.).
Code is located here.
On 5/28/07, Roberto Fichera [EMAIL PROTECTED] wrote:
At 19.19 28/05/2007, you wrote:
On 5/28/07, Roberto Fichera mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
At 17.09 28/05/2007, Richard Alam wrote:
Yes, we have some downloadable code. We are in the process of completing
the instructions
From: EdPimentl [EMAIL PROTECTED]
Date: Sun, 27 May 2007 16:12:09 -0400
There will be a number of companies set to offer similar services.
In 3 months we will have a 24 port SIP-GSM-SKYPE gateway
-E
On 5/27/07, Dean Collins [EMAIL PROTECTED] wrote:
I was cleaning through some old IT
Luki wrote:
Perhaps a naive question, but how does 0.137% CPU utilization per call
equal 1735 MHz per call?
If 1735 MHz / 0.137% = 1735 MHz / 0.00137 = 1266423 MHz at 100%
utilization ??! Even with 4 CPUs, those would be 316 GHz CPUs.
I think you meant:
Average CPU utilization per call: 0.137%
Hello,
PBX vendors used to sell software extensions providing enterprise services
to cell phones.
Main features were :
- 4 digit dialing or directory access,
- call forwarding,
- unified messaging.
Now that WiFi and dual mode cell phones get more popular, these Java-based
software should be
JR Richardson wrote:
Do you get any errors at max call capacity about too many open files? You
may try increasing your file descriptors.
JR,
Thanks for the response, but I have the maximum number of open files
available to Asterisk set to 65536.
Thank you,
Matthew Roth
InterMedia
At 19.56 28/05/2007, you wrote:
On 5/28/07, Roberto Fichera mailto:[EMAIL PROTECTED][EMAIL PROTECTED]
wrote:
At 19.19 28/05/2007, you wrote:
On 5/28/07, Roberto Fichera mailto: [EMAIL PROTECTED]mailto:[EMAIL
PROTECTED][EMAIL PROTECTED] wrote:
At 17.09 28/05/2007, Richard Alam wrote:
Yes,
William Moore wrote:
Are you recording memory figures as well and have you checked the
total used memory? Or did I miss it somewhere? Thanks for doing
this, scalability testing is always good.
William,
This round of benchmarking is heavily focused on CPU utilization,
because it is causing
Hi Olivier,
Do a search on my blog www.collins.net.pr/blog for Orative as a
suggested application.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
JR Richardson wrote:
The Dual-Core system you are working with must have cost a bundle, several
thousand. My approach has been to stick with single cpu, single core
servers and add more servers to the cluster, versus building bigger, faster
Proc servers. With sub $1000 servers, I can achieve
Mark Coccimiglio wrote:
Sounds like you are running into the hardware limitations of your
systems PCI or Front Side Bus (FSB) and not necessarily an issue of
asterisk. In short there is a limited amount of bandwidth on the
computer's PCI Bus (33 MHz) and the FSB (100-800MHz). One thing to
Hi Sebastien,
I'm just a lowly user but I will tell you what I think I understand about
it.
There is nothing in the Octasic documentation that suggests you can have
continuosly updated statistics but I agree that would be a nice to have
feature. Have you tried contacting Octasic about that?
Sorry but I must have missed it if someone else responded. If the built in
fax reception doesn't work very well what about the 3rd party stuff
mentioned on the Asterisk Wiki?
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Monday, May 28, 2007 8:38 AM
To: Asterisk
I was wondering whether someone could help me. I installed sipp on my Asterisk
server but I don't really understand how does it fonction! Has someone ever
tried it?If you can explain to me the principle, I would be extremely
grateful.Thank you very much in advance.
Lancez des recherches
I am still having issues with my Polycom 301 phones when I disable DHCP. I
give the phone a static address and I keep getting the error 'could not
contact boot server using existing config'. As soon as I set it back to
DHCP enabled the phone can see the boot server and I'm online.
Steve
Doug Lytle wrote:
shadowym wrote:
So what is the bottom line? Does it work or not. I've heard stories it
As it has been said many many times before, Fax detection is an art
and most of the time is not reliable. Faxing on the other hand, using
iaxmodem along with HylaFAX+ works very
I gave up on the rxfax business as it never worked for me. I use iaxmodem and
hylafax and it works perfectly, every single time i use it. inbound or outbound
doesnt matter.
I have not read about anyone using iaxmodem and hylafax having any issues. and
its fairly easy to setup. Took me about 1
On 5/24/07, shadowym [EMAIL PROTECTED] wrote:
So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.
What's the bottom line with recent updates on 1.2.x? Is it
http://www.thecoccinella.org/ looks pretty nice
I have not tried this one. It has been a couple of years since I played
around with IM clients and I cannot remember what I was using.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com/
KB3OPB
_
From: [EMAIL PROTECTED]
Sounds like a firmware bug, VLAN or other network configuration bug in
the phone (subnet perhaps?)
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forum
Sent: Monday, May 28, 2007 3:56 PM
To:
Please qualify your usage. A couple faxes a day, a couple hundred, a
couple thousand, or a couple hundred thousand?
Are you running asterisk and hylafax on the same machine? What is your
TDM connectivity?
Hylafax uses quite a lot of CPU juice. Anyone ever scale up a quad
T1/E1 server
Anyone know how to send a post transfer (and possibly post hangup) message to
an aastra set with the pickup extension
Dave
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]
For those who just want it to work...
Giving you complete IT peace of mind.
(Sent via Blackberry - hence
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Yes, we are looking for that. Do you know of any projects that provides
those? I know one written in TCL/TK.
You might also want to have a look at
http://www.version2software.com/v2whiteboard.html - its a plugin for the
Java based Jabber client
Quoting Steve Totaro [EMAIL PROTECTED]:
Please qualify your usage. A couple faxes a day, a couple hundred, a
couple thousand, or a couple hundred thousand?
well what is your usage where it doesn't work ?
I would like to know where it does and doesn't work as well but so far
various
I am having a problem setting the default language for Zap interfaces.
I have an Asterisk 1.4.4 server on CentOS 5 with two Astribank 8 units
for analog devices. Here is a sample configuration on one of the ports:
language=es
context=oficina
callerid=Miriam Perez Vite100
mailbox=100
On Mon, May 28, 2007 at 02:22:34PM +0530, ram wrote:
Hi
I have 100XP Digium clone card
Installed in my pc
and compiled zaptel and asterisk again after installing the card
but after i rebooted
i can load zaptel and wcfxo modprobe with out any problem
but when i intiated ztcfg
I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no
baby sitting, I receive about 20 and it requires no baby
sitting
Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and
hylafax lists for much bigger examples
Cheers Duncan
-Original
On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote:
As you can see I set the language=es parameter (and do this for all
interfaces). I installed the spanish sound set for Asterisk
in /var/lib/asterisk/sounds/es (with links to the appropriate
directories for letter, digits,
Steve Totaro wrote:
Please qualify your usage. A couple faxes a day, a couple hundred, a
couple thousand, or a couple hundred thousand?
Couple hundred thousand per month - at least on one installation.
Are you running asterisk and hylafax on the same machine? What is
your TDM
On Tue, May 29, 2007 at 12:34:33AM +0300, Tzafrir Cohen wrote:
On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote:
As you can see I set the language=es parameter (and do this for all
interfaces). I installed the spanish sound set for Asterisk
in /var/lib/asterisk/sounds/es
Thanks for all the replies. Seems there are at least 2 or 3 people giving
strong recommendations to iaxmodem/hylafax as a reliable (ie. business grade
production) solution. That is just the sort of feedback I was looking for.
My application is just standard business reception of faxes. Right
Thanks Stefan! I was just thinking the other day that it would be great
if I could whiteboard in Spark.
Back on topic, I'm definitely interested in this web conferencing app.
I'll have to check it out once a .war is made available and I have a few
spare moments.
- Brad
-Original
On Mon, May 28, 2007 at 07:18:59PM +0530, rajesh koniki wrote:
Hi,
I was looking for a way to pass alphanumeric variables to asterisk via
the keypad, found this application app_dtmftotext.c , And I already tried
with 'spandsp' application for this. But I am getting errors.[I followed
the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hey Brad,
I am not sure if you know about the Asterisk-IM plugin for Openfire.
Basically it supports dialing contacts and arbitrary numbers through
Spark and updates presence based on being on call or not.
One of our next steps would be to integrate
Yes that makes more sense. Now to the problem, please post your
zapata.conf as well as your zaptel.conf. Also if you don't mind
downloading the config file from the Panasonic TD1232 and email to me
off list so I can take a look at it and make sure the settings are ok
on the panasonic side.
Thank
On Tue, 2007-05-29 at 00:53 +0300, Tzafrir Cohen wrote:
On Tue, May 29, 2007 at 12:34:33AM +0300, Tzafrir Cohen wrote:
On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote:
As you can see I set the language=es parameter (and do this for all
interfaces). I installed the
Greg Kennedy wrote:
I gave up on the rxfax business as it never worked for me. I use
iaxmodem and hylafax and it works perfectly, every single time i use
it. inbound or outbound doesnt matter.
I have not read about anyone using iaxmodem and hylafax having any
issues. and its fairly easy to
A colleague of mine did some testing the other day, with a digium TDM400
with FXS modules hooked up to fax machines and a TE120P hooked up to our
testing E1 line.
It seems to work pretty well, and she said it was easy to configure.
PaulH
On Mon, 2007-05-28 at 14:55 -0700, shadowym wrote:
Tim Litwiller wrote:
Greg Kennedy wrote:
I gave up on the rxfax business as it never worked for me. I use
iaxmodem and hylafax and it works perfectly, every single time i use
it. inbound or outbound doesnt matter.
I have not read about anyone using iaxmodem and hylafax having any
issues.
If you are a junk spam faxer then it should suit your needs.
If you occasionally send faxes and if you do not receive one or the
other party does not receive one or it spits out junk but that is OK,
then it should fit your needs.
If you are faxing contracts or other important documents that
Chris Earle wrote:
Hi all,
Years ago, I was pretty sure attempting to use two TDM400p cards in one
machine was recommended against by Digium ... probably because the cards
couldn't hack it, and/or interrupt problems etc
I have seen some posts recently that seem to indicate it is in fact
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Monday, May 28, 2007 8:57 PM
To: Chris Earle; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Multiple TDM400p cards in one
Quoting Steve Totaro [EMAIL PROTECTED]:
If you are a junk spam faxer then it should suit your needs.
If you occasionally send faxes and if you do not receive one or the
other party does not receive one or it spits out junk but that is OK,
then it should fit your needs.
If you are faxing
According to your CLI output, the channel is being torn down.
Is there a lag on the CLI between the inside channel and the outfacing
channel getting the hangup request?
Does this only happen on mobile phones? I know if I call my cell and
hangup, it will continue to ring a couple or even a few
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Pounder
Sent: Monday, May 28, 2007 9:10 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] RE: Bottom line on fax reception
Quoting Steve Totaro [EMAIL
Quoting Steve Totaro [EMAIL PROTECTED]:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Pounder
Sent: Monday, May 28, 2007 9:10 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] RE: Bottom line on fax reception
Let me further qualify my results. This was done with whatever the
current stable versions of Asterisk, Hylafax, and IAXmodem were
available in January of this year. The faxes were outbound. PDFs
put
into a Samba share and a cron job moving them over to the Hylafax
monitored
Hello,
Sometimes, when a call comes in from the PSTN through our VoIP gateway,
the information that is sent to our web page that logs calls includes the
original CID name instead of the one that is we expect to be rewritten on
the fly using Asterisk's LookupCIDName:
=
returning newbie.
Trying to register ekiga for the first time to my asterisk server only.
[204]
user=204
context=internal
type=friend
secret=xxx
insecure=very
canreinvite=no
host=dynamic
disallow=all
allow=ulaw
allow=alaw
nat=no
Can anyone tell me what I am missing?
I am not
Hi all:
We are looking for someone with experience in Alcatel PBX - PRI -
Asterisk integration
Please get in touch off list.. We're wanting to hire a professional
subcontractor, developer or company to get around some issues like these:
Asterisk shows PRI to Alcatel is up, but when
Hi,
You need to enable overlapdial.
Regards,
Sahil Gupta
Chief Executive Officer
VoiceValley Group of Companies
Phone: +61-7-30188403
Fax: +61-7-30188499
On Tue, 29 May 2007, Carlos Hernandez wrote:
Hi all:
We are looking for someone with experience in Alcatel PBX - PRI - Asterisk
On Mon, May 28, 2007 at 06:08:38PM -0500, Carlos Chavez wrote:
On Tue, 2007-05-29 at 00:53 +0300, Tzafrir Cohen wrote:
On Tue, May 29, 2007 at 12:34:33AM +0300, Tzafrir Cohen wrote:
On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote:
As you can see I set the
On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote:
returning newbie.
Trying to register ekiga for the first time to my asterisk server only.
[204]
user=204
context=internal
type=friend
secret=xxx
insecure=very
canreinvite=no
host=dynamic
disallow=all
204/20466.176.193.46D 5063 Unmonitored
It just came up after a reboot on its own???
Go figure, windows problem!
Thank you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Tuesday, May 29, 2007
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