Hi,
Our asterisk installation will be a man-in-the-middle providing local,long,international VOIP services to our customers and our asterisk will be connect via VOIP to international carriers.
We use asterisk 1.2.5 with mysql in centos 4.2 Kernel 2.6
I have looked at astbill and it sounds
Where can I do a keyword search of the posting in biz and users forums? asterisk.org just links to http://lists.digium.com/pipermail/and that doesn't let me do a string search across all postings.
thanks,
-- ---Erick PerezLinux User
Superb replies.
Thanks to Jon and Noah
On 3/27/06, Noah Miller [EMAIL PROTECTED] wrote:
Hi Erick - Where can I do a keyword search of the posting in biz and users forums?
asterisk.org just links to http://lists.digium.com/pipermail/ and that doesn't let me do a string search across all
Hi, I use Centos 4.2 (kernel 2.6) on an Intel P4 at my house and i plan to use that machine to build asterisk.
However the target system (a small test server) is a latest Intel Celeron 2.x Ghz but with Centos 4.2 kernel 2.6.
I don't build things a lot (thanks for the RPMs) so i better ask before
On 3/27/06, Erick Perez [EMAIL PROTECTED] wrote:
Hi,
Our asterisk installation will be a man-in-the-middle providing
local,long,international VOIP services to our customers and our asterisk
will be connect via VOIP to international carriers.
We use asterisk 1.2.5 with mysql in centos 4.2
,
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http
: chan_h323.so: load_module
failed, returning 1
Mar 29 21:59:41 WARNING[6531] loader.c: Loading module chan_h323.so failed!
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, 2006 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] welltech Wellgate 3804 in SIP mode
On Mar 25, 2006, at 6:26 PM, Erick Perez wrote:
Martin, i guess im in dumb mode today because i don't get what you say, may
also be because
dial number twice, so this is
called two-stage dialing.
artifex
On 3/31/06, Erick Perez [EMAIL PROTECTED] wrote:
one-stage calling function?
On 3/30/06, kevin ling [EMAIL PROTECTED] wrote:
Yes,
Same configuration as Martin.
1.for incoming call just set the 3804 hotline to one
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root root 0 Apr 1 20:32 /var/run/asterisk.ctl
Suggestions?
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[EMAIL PROTECTED] wrote:
the user you are connecting as should have full rights to /var/run/asterisk:
http://www.voip-info.org/wiki-Asterisk+non-root
hth
-Original Message-
From: Erick Perez [mailto:[EMAIL PROTECTED]
Sent: Monday, April 03, 2006 9:28 AM
To: Asterisk Users Mailing List
Thanks Steven, I was building an RPM and forgot that.
/etc/asterisk.conf was:
astrundir = /var/run
now is:
astrundir = /var/run/asterisk
Works perfectly.
thanks,
On 4/3/06, Erick Perez [EMAIL PROTECTED] wrote:
I ran strace and got this:
open(/var/run/asterisk.pid, O_WRONLY|O_CREAT
and
hear myself speaking, I use logitech headsets for the operation.
I have also played with rx/tx gain values from the default of 0.0 all
the way to 8.0
suggestions?
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required
by business and enterprises.The Audio Conference Bridge developed is
superior in voice quality among other Audio Conferencing Bridges in
the industry.
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Does Asterisk support a Brooktrout TR1000 ?
http://www.cantata.com/products/tr1000/
It seems that they have linux drivers.
I have one around and was wondering if it works with *.
Thanks,
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To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and Brooktrout TR1000
http://www.asterisk.org/hardware
Erick Perez wrote:
Does Asterisk support a Brooktrout TR1000 ?
http://www.cantata.com/products/tr1000/
It seems that they have linux
sucess stories with:
AudioCodes MP124-C/FSX/AC/SIP ---Asterisk---internet---Vonage setups?
Thanks,
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file via the Embedded Web Server, refer to
Section
5.9.2.1 on page 120.
On May 24, 2006, at 3:00 PM, Erick Perez wrote:
Just a question, has anyone knows how to blank or factory
reset an
AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit).
I purchased them second
Has anyone run asterisk with SYSMASK?
http://wims.unice.fr/sysmask/doc/
or maybe securing asterisk installations have another howto anywhere?
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will be nice.
Comments?
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and features.
Comments are welcomed,
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When things from MANTIS are merged into stable asterisk. Where does it
says that?
On 5/25/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Erick Perez wrote:
does anybody knows if this patch made it into Asterisk Business Edition?
http://bugs.digium.com/view.php?id=4825
ABE never includes any
many thanks to all.
On 5/25/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Erick Perez wrote:
When things from MANTIS are merged into stable asterisk. Where does it
says that?
In Mantis, and in the commit message for the Subversion repository.
But the patch you are asking about
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should I use mpg123 with asterisk 1.2.7 or should i use the native
player asterisk has?
the target machine will receive heavy load.
also, has anyone succedded in compiling mpg123 in a dual core pentium
with centos 4.3 ?
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i found it sorry.
:((
On 5/26/06, Erick Perez [EMAIL PROTECTED] wrote:
Hi,
I am ready to compile asterisk in centos 4.3 x86_64 however i cannot
find the kernel-sources package (rpm). Anyone on this list that can
point me as to where is it? Im sure im looking in the wrong
mirros.centos.org list
Thanks to all. Native format will be.
On 5/27/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
Vahan Yerkanian wrote:
Erick Perez wrote:
should I use mpg123 with asterisk 1.2.7 or should i use the native
player asterisk has?
the target machine will receive heavy load.
mpg123 was used back
? Transcoding
MP3s but sending a single stream or separate streams per call under
native?
When I say high, I mean 1,000+ calls.
Thanks,
Steve
Erick Perez wrote:
Thanks to all. Native format will be.
On 5/27/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
Vahan Yerkanian wrote:
Erick Perez
under a load of sixty channels, I can
confirm that for sure.
Thanks,
Steve
Erick Perez wrote:
Interesting.
So, i will have to test then...
On 5/27/06, Steve Totaro [EMAIL PROTECTED] wrote:
In my very limited testing of native, each channel was receiving a
different stream (each caller heard
processes, I figured
native was not the way to go since all of the different audio streams.
Mpg123 works perfectly for me under a load of sixty channels, I can
confirm that for sure.
Thanks,
Steve
Erick Perez wrote:
Interesting.
So, i will have to test then...
On 5/27/06, Steve Totaro [EMAIL
I have the following extension to dial outside via SIP
it's like this:
phoneasterisk-internet-SIP providerUSA
exten = _91NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN},55,o)
exten = _91NXXNXX,3,Hangup
I want to strip the digit 9
me dumb, a quick question slipped my mail.
should google more next time.
Thanks to all.
On 5/30/06, C F [EMAIL PROTECTED] wrote:
It's all in README.variables or here:
http://voip-info.org/wiki/view/Asterisk+variables
use ${EXTEN:1}
On 5/30/06, Erick Perez [EMAIL PROTECTED] wrote:
I have
?
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mailing list
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will play along soon, so you'll have a CRM and a
VOIP server working together.
Any other features?
Feel free to ask. or email me directly.
Erick Perez
eaperezh ((at)) gmail ((dot)) com
.
On 5/31/06, Kit Gerrits [EMAIL PROTECTED] wrote:
Hello all!
Please forgive me if I am asking stupid
Hi, does anyone know the prices for g729 codecs from Digium? I sent an
email a while ago to them but haven't got any response so far.
Prices are per unit or volume?
Thanks,
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incoming calls?
thanks,
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While sending calls to a SIP provider, the following warning generates:
-- Executing Dial(SIP/1000-c317,
SIP/[EMAIL PROTECTED]:5060|55|o) in new stack
-- Called [EMAIL PROTECTED]:5060
-- SIP/209.120.202.94:5060-0533 is making progress passing it to
SIP/1000-c317
--
a conference with other users
also at g729 asterisk will not transcode (when using app_conference)?
Thanks,
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-constant
# this is fun for PPC
#CFLAGS += -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic
# this is fun for x86
# The line below was commented by Erick Perez [EMAIL PROTECTED]
#CFLAGS += -march=pentium3 -msse -mfpmath=sse,387
# adding -msse -mfpmath=sse has little effect.
#CFLAGS += -O3 -msse
. Fleming [EMAIL PROTECTED] wrote:
- Erick Perez [EMAIL PROTECTED] wrote:
Or if i have SIP/g729 users and i create a conference with other
users
also at g729 asterisk will not transcode (when using app_conference)?
It is not possible to mix conference audio together without converting
-funroll-all-loops -fprefetch-loop-arrays
-fsingle-precision-constant
# this is fun for PPC
#CFLAGS += -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic
# this is fun for x86
# The line below was commented by Erick Perez [EMAIL PROTECTED]
#CFLAGS += -march=pentium3 -msse -mfpmath=sse,387
# adding
They key point is to disable de x86 CFLAGS and add this one
CFLAGS += -march=k8 -fPIC
k8 is the machine type for x86_64
On 6/4/06, Erick Perez [EMAIL PROTECTED] wrote:
This is my makefile, it compiled ok. I will test it tomorrow but if
you have somewhere to test today, let me know.
# $Id
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one meetme bleeding
into another, but since you mention meetme and call center I assume
you are using VICIDIAL which might mean you are having some issues
with your agent's not logging out properly.
MATT---
On 6/5/06, Erick Perez [EMAIL PROTECTED] wrote:
Have anyone experienced mixed meetme
planning for a voicemail server.
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Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
mailto: [EMAIL PROTECTED]
mailto: [EMAIL PROTECTED
?
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Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
mailto: [EMAIL PROTECTED]
mailto: [EMAIL PROTECTED
i will check on SER in redirect mode.
I found
http://developer.berlios.de/projects/ser
www.iptel.org/ser
http:/openser.org
Will check on the differences.
Thanks
On 6/7/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
Erick Perez wrote:
If you have the need to implement 5 asterisk boxes
,
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: 5338.11
clflush size: 64
cache_alignment : 128
address sizes : 36 bits physical, 48 bits virtual
power management:
--
Erick Perez
When is supposed to be released the Zimbra+Asterisk version?
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Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
mailto: [EMAIL
,
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Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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Wow, I missunderstood a zimbra article I read somewhere.
A zimletthanks.
On 6/9/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Erick Perez [EMAIL PROTECTED] wrote:
When is supposed to be released the Zimbra+Asterisk version?
There is no 'Zimbra+Asterisk' version of anything
for
--
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
.
On 6/12/06, BJ Weschke [EMAIL PROTECTED] wrote:
On 6/12/06, Erick Perez [EMAIL PROTECTED] wrote:
I have this server I need to put to work.
The option I have is to make it work as a small office PBX with SIP
users and a Digium E1 Card for PSTN service.
24 SIP users and one E1 card in an Intel
(bus bandwidth table)
http://www.acme.com/build_a_pc/bandwidth.html
http://www.lostcircuits.com/memory/ddrii/
http://bvio.ngic.re.kr/Bvio/index.php?title=Front_side_bus
On 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Erick Perez wrote:
I just don't want to install it and then after a 5th user
and if it doesn't work try a better box. You
already have the cheap machine, and the card will remain the same
regardless
of what box you use.
-Original Message-
From: Erick Perez [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial
To UNSUBSCRIBE or update options visit:
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Panama Sistemas
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Panama, Republica de Panama
Cel Panama. +(507
.
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units.
Thanks,
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Panama, Republica de Panama
Cel Panama. +(507
want to do provisioning automatically.
in the 46xxsettings.txt file there are no such parameters
thanks,
--
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel
Hi, my employer will send me to Belgium, Bruselles from Nov 11 to Nov
26. Will an asterisk meeting take place anytime between those dates?
Since in my country we have *no* meetings whatsoever I was wondering if
there was going to be any.
Or a tech fair showing asterisk. Or maybe a local asterisk
, I just want some links to AMP modules and see if i
can build from different sources a graphical interface that does
end-user pbx functions
BTW call accounting and billing will be nice too.
Thanks,
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Linux User 376588
http://counter.li.org
.
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Update to myself:
So in terms of equipment I will need:
Sangoma a102 E1 (two E1 ports) plus a E1 crossover cable
a channel bank with 8 FXS ports
sounds expensive for just 8 analog ports. Any ideas?
On 8/31/05, Erick Perez [EMAIL PROTECTED] wrote:
Hi, Im about to start shopping parts
, 2005-08-31 at 18:37 -0500, Erick Perez wrote:
Update to myself:
So in terms of equipment I will need:
Sangoma a102 E1 (two E1 ports) plus a E1 crossover cable
a channel bank with 8 FXS ports
sounds expensive for just 8 analog ports. Any ideas?
On 8/31/05, Erick Perez [EMAIL
to
do separate VLANs for any SIP hard phones you deploy. This adds another
layer of security and reliability.
Hope this helps.
On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote:
-M, The norstar has no E1 card, i will have to ask the nortel provider
for the cost
by another device such as your
PIX. You might want to check with the ADSL router manufacturer just to
be safe.
On Thu, 2005-09-01 at 09:25 -0500, Erick Perez wrote:
Do i have to change the adsl routers? or just do QoS with the Layer 3
switches?
Will my ADSL router respect the QoS setting
.
Something else to keep in mind.
best of luck.
On Thu, 2005-09-01 at 22:03 -0500, Erick Perez wrote:
Why an L3? just for the QoS part?
I checked the alliedtelesyn 8624T at $1000.00
http://www.cdw.com/shop/products/default.aspx?EDC=772793
but i also looked at the 8550T which has 48
:
Erick Perez wrote:
So, with this i solve the issue on main office. But what about the two
remote? they are so little that they will not let me place another *
box there. The phones will be SIP and they are like this
INTERNET--PIX--LAN(machines and sip phones). The pixes in those two
offices have
channel bank is about USD 1500 + a t1 card USD
500 + a USD 1000 computer = 3 thousand us dollars + my installation
fees (life isn't free).
Sounds expensive for such a small install.
Suggestions?
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Thursday, September 08, 2005 8:27 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Solution for 12 to 16 FXO to asterisk
connection
Hi, today
Using sipura sip/g729 to connect to an asterisk server that will
server as a gateway to a VOIP provider, all in g729 will require to
purchase codecs from Digium?
also, in this scenario the transcoding is almost non-existent right?
I have read many documents about the type of codecs, and g729
anyone with some info on this?
thanks again.
On 9/14/05, Erick Perez [EMAIL PROTECTED] wrote:
Using sipura sip/g729 to connect to an asterisk server that will
server as a gateway to a VOIP provider, all in g729 will require to
purchase codecs from Digium?
also, in this scenario
Perez [EMAIL PROTECTED] wrote:
anyone with some info on this?
thanks again.
On 9/14/05, Erick Perez [EMAIL PROTECTED] wrote:
Using sipura sip/g729 to connect to an asterisk server that will
server as a gateway to a VOIP provider, all in g729 will require to
purchase codecs from
Hi guys, given 1.2.1 is out. How is the t38/fax support going on?
also, can someone point me to proven brands/configs with ip fax capable
machines? Fax machines with a lan port (i heard of them but havent
found them online).
or a fax machine plugged to a converter that actually works for heavy
Existe un Debian 3.1 y un Ubuntu 5.1. Si hay un Debian 1:3.3.5-8ubuntu2 debe ser algo muy viejo.
Ademas, te comento que esta lista es en ingles. Trata de postear en ingles.
This list is in english. Please try to post in that language so more people can help you.
On 12/20/05, Will Velez [EMAIL
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The cyber-telecom is cheaper compared to the doc-n-talk unit.
http://cyber-telecom.net/store/product_info.php?cPath=1products_id=29
however, they both work for 800/900/1800 bands.
Any products to work in the GSM 850mhz arena?
In our country, GSM runs 850mhz.
Brian: Do you need fancy features or
BTW
Places that sell CellSockets that are know to work.
http://www.cyber-telecom.net/store/900/1800 GSM. No phone needed just SIM card.
http://www.cellantenna.com/Dockingstations/cellsocket.htmcellular phone accessory that allows you to dock your cellular phone and integrate it with your Land
I just found a Dialogic VFX/41JCT-LS (4 analog ports) in a drawer. I can use it in my house with asterisk at home project.Can I use that with asterisk?Where can I download proper drivers?--
---Erick PerezLinux User 376588http://counter.li.org/(Get
Well, this product from signate uses infiniband...
It has 4 slots for quad e1/t1 per slot.
http://www.signate.com/pdf/TelephonyServer.pdf
Just read the PDF. Obviously this is not an x86 Pc. I wonder if you want to build your own or were looking for a beast like this.
BTW, any real world comments
, outbound, inboud, voicemail,fax,cell phones, etc.
So far, shame on me, I have no idea where to start in terms of equipment.
Or I can go out there and buy a 20k machine(s) to run 4or5 E1sIt
will run, but I will never learn why.
thanks,
--
---
Erick
Hi,
emails to astricon.net seems to bounce (at least for me)
I need information about proper authorized Asterisk training in the
Miami, FL area and the possibility of later DCAP testing.
Thanks,
--
---
Erick Perez
Linux User 376588
http://counter.li.org
was trying
to use other people's configs have others help me etc.
thinking it was the short way out. It ended up being
the long way. The best thing is to learn it on your
own. Just my $0.02 .
Dovid
--- Erick Perez [EMAIL PROTECTED] wrote:
Hi,
emails to astricon.net seems to bounce (at least
Is this an outgoing maintain? will you keep rebuilding them?
On 1/19/06, Vladimir Montealegre [EMAIL PROTECTED] wrote:
yes, asterisk work with centos
- Original Message -
From: Eric Bishop
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, January 19, 2006
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
---
Erick Perez
Linux User 376588
http://counter.li.org/ (Get counted!!!)
Panama, Republic of Panama
___
--Bandwidth and Colocation provided
nobody uses avaya phones with asterisk?
On 6/20/06, Erick Perez [EMAIL PROTECTED] wrote:
Hi, I setup my tftp to send SIP configurations (the bin files) to the
avaya phone. When it finished loading and rebooting it asked for the
extension and the password and the asterisk ip address. I had
the Kettle Black.
Good phone, great sound, just no support and a bit wonky on the features.
My 2 cents.
On 6/21/06, Erick Perez [EMAIL PROTECTED] wrote:
nobody uses avaya phones with asterisk?
On 6/20/06, Erick Perez [EMAIL PROTECTED] wrote:
Hi, I setup my tftp to send SIP configurations
.
On 6/21/06, Erick Perez [EMAIL PROTECTED] wrote:
Thanks for your comments Tom. Indeed the MWI and the programmable
buttons are the only things that do not work for me. Besides that, the
phone is great and the audio quality is superb.
Did you managed somehow to make the MWI work?
Will keep
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