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-30 seat range, with 50-70 agents on shifts. Nearly everybody seems
to be quite happy with what they've got, at least since * hit 1.0.
Cheers,
l.
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Hi there, what phones are available that have two ethernet ports?
I want to do some cabling at a new installation and i heard there are
such phones (SIP i guess) out there. That way i dont have to run two
cat5 to the user desktop.
I think 3COM had one but can't find the web site reference for the
Update:
found the 3Com® 3101 Basic Speaker Phone
Provides dual port 10/100 switched Ethernet for one-wire connectivity
between the phone and a PC
any others not so expensive? does these 3com sip phones work with * ?
On Sun, 2 Jan 2005 16:35:12 -0500, Erick Perez [EMAIL PROTECTED] wrote:
Hi
or update existing wikis
and voip-info.org (when we all agree of course).
I bought the * book from signate. It's very good but lack some
things...I'm waiting for the next revision (if any).
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Erick Perez wrote:
This question is for my own knowledgei have no experience on this
electrical area.
why do you want to run -48vdc equipment? what's the advantage of doing that?
On Tue, 18 Jan 2005 13:58:59 +0100, Daniel Nyström
[EMAIL PROTECTED] wrote:
What
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as traffic
grows and tested/used with Asterisk before?
commercial or open source links are ok.
thanks,
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code. I didn't
think it was too bad.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Tuesday, January 25, 2005 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] New ip billing solution
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im sorry insmod
On Thu, 27 Jan 2005 09:22:11 -0500, Erick Perez [EMAIL PROTECTED] wrote:
if it is on Linux hardware with * you'll need to get your hands on the
Linux drivers for your dialogic board which are not publicy accesible
(or are they?).
You must have it recognized by the linux
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, voicemail
SIP-TDM communications, voicemail
how may users (SIP hardphones and analog phones via CPE equipment)
Thanks,
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Is there an * patch repository i'm not aware of? or do i have to
google the web or ask here?
also, are these patches for old * versions? does * 1.06 needs them?
Many thanks in advance,
Erick.
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Sirs, i just joined the mailing list and i have a question:
What kind of phones can be used with asterix (phones with screen). Basically
to see whos calling, display the time,etc...Just like normal phones with
display screen do.
Thanks,
Erick
___
Given the myriad of telehpone cards available I like to ask this forum for
the following combination:
Asterix on Linux redhat (9.0 or Fedora)
10 analog extension using conventional phones (lets say Panasonic kx-ts3
analog)
4 analog lines coming from our telco
So i will need 3 TDM40B (total 12
I read in the archives a post from last year about the Dialogic drivers not
being free for use with Linux/Asterisk.
So, I have a VFX/41JCT-LS to try with *
Suggestions? Purchase digium boards is not an option. We want to test the
app before buying any other hardware.
thanks,
erick,
, 2004-06-18 at 15:39, Erick Perez wrote:
I read in the archives a post from last year about the Dialogic
drivers not being free for use with Linux/Asterisk.
So, I have a VFX/41JCT-LS to try with * Suggestions? Purchase digium
boards is not an option. We want to test the app before buying any
ok Steven, so i dump dialogic. but my question remains. Are there any
free-available linux drivers for the * pbx/dialogic or do i really have to
dump my card.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Friday, June 18, 2004 4:22 PM
Hi people,
question one
i see that asterisk is now in 1.x release. having tried it in the past
i want to know if i can use a voice modem as an outgoing line.
i know in the past that was not possible/supported so im just asking
in case the option is now available.
question two
im planing to use
I see that the 100p is a modem with an Ambient chipset. Why does it
sell for 80$ in some places? i can get Ambient pci modem down here for
9 dollars. Any difference?
On Tue, 7 Dec 2004 10:58:55 +, Jon Lawrence [EMAIL PROTECTED] wrote:
On Tuesday 07 December 2004 04:36, Erick Perez wrote
Hi, i just installed latest asterisk on fedora rc2 and on the same
machine i installed a sip soft phone called kphone. Kphone complains
about /dev/dsp being used and can't place/answer calls (/dev/dsp is
obviously used by asterisk) . how can share my sound card with these
two programs?
or
can i
Hi, the following logs are being generated while i test sip-to-sip
windows software phones.
Dec 7 17:05:16 WARNING[-159503440]: chan_sip.c:683 retrans_pkt:
Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102
(Critical Request)
== No one is available to answer at this time
Dec 7
Hi,
While I wait for my unresponsive telco to provide some assistance, can
you provide some configuration details for the following config?
Sangoma 102 (dual E1) card
Location: Panama, Central America
Telco: Cable Wireless Panama
Lastest stable asterisk 1.2.x compiled from sources
Site A in one
I have received the follwing info from my telco.
E1, PRI, CAS, HDB3, dss1
any help?
On 7/25/07, Erick Perez [EMAIL PROTECTED] wrote:
Hi,
While I wait for my unresponsive telco to provide some assistance, can
you provide some configuration details for the following config?
Sangoma 102 (dual
= 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 1
TDMV_DCHAN = 16
[w1g1]
ACTIVE_CH = ALL
TDMV_ECHO_OFF = NO
TDMV_HWEC = YES
thanks for your help.
--
Erick Perez
,Answer()
exten = 12345678,2,Playback(Welcome)
...
12345678 = The DID number you are calling to reach E1
Idris
-Original Message-
From: Erick Perez [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 26, 2007 7:03 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk
for your kind and prompt help.
On 7/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
If you do not have any alarms and PRI debug span 1 still gives you
nothing then you need to call your telco and say I'm not getting any
Q.931 messages on the D-Channel.
Stephen Bosch wrote:
Erick Perez wrote
does not requiere to use an AC adapter if used with
PoE injectors/switches.
Can non-Cisco PoE injectors/switches be used with this phone?
Thanks,
--
Erick Perez
, whether it is
Callmangler or Asterisk or whatever. If you buy one used, then you need
to pay to re-license it as well.
The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you
will need a switch that provides Cisco PoE for it to work.
Erick Perez wrote:
Hi there,
In Cisco web
.
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On 1/23/06, Facundo Ameal [EMAIL PROTECTED] wrote:
Erick Muchas Gracias por la respuesta.
I'm not using any of that projects, it's my own Asterisk installation
onto slackware 10.
well what can you tell about sipura ones?
2006/1/23, Erick Perez [EMAIL PROTECTED]:
Hola Facundo, saludos desde
Besides the codecs that * supports. Is there any ISAC implementation
for asterisk available?
This is to be used mainly with softphones, i haven't seen any
hardphones that support this codec.
Thanks,
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Well, skype. but i was tweaking some code. This is more a question for
lab usage.
On 1/28/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
Erick Perez wrote:
Besides the codecs that * supports. Is there any ISAC implementation
for asterisk available?
This is to be used mainly
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asteirsk PBX.
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/claro91/66944780) in new
stack
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/6000-0820e870,
SIP/claro91/66944780) in new stack
-- Called claro91/66944780
[May 13 17:37:40] WARNING[5522]: chan_sip.c:11860
handle_response_invite: Received response: Forbidden from 'Erick
Perez sip:[EMAIL PROTECTED];tag
,
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I realized that queuemetrics uses Java.
Is java available as an rpath package or do I need to get it from sun?
Also, will it break asterisknow?
Thanks.
On 5/21/07, Erick Perez [EMAIL PROTECTED] wrote:
Can I use queuemetrics with asterisknow?
I mean, if I modify the dialplan to use queuemetrics
analog_lines_fxo_signal--(FXS side)FXStoSIPdevice(SIP
side)--Asterisk
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Thanks Jerry. Are the avaya station ports a special type ?
On 1/18/07, Jerry Jones [EMAIL PROTECTED] wrote:
Connect to the avaya line ports, not station ports.
On Jan 18, 2007, at 10:46 AM, Erick Perez wrote:
Hi, this is a signalling question:
I have a 4port fxs-to-sip where i connect
remember how to factory reset them and what
will be the default password in the GUI.
thanks for your help.
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Panama Sistemas
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Cel Panama
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from autosupport. It is available at:
http://pastebin.com/868590
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both not available.
but thanks.
On 1/28/07, Leif Neland [EMAIL PROTECTED] wrote:
Erick Perez wrote:
Hi there, im looking for another place that provides manuals and
firmware updates for the ATCOM AT 468 and their configuration with
asterisk.
the site www.atcom.com.cn has non functional
I have tried compiling asterisk with -march 586 and 386 and the
deadlocks minimizedin 386 but did not dissapear.
Is this because of asterisk, my epia or centos?
On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote:
In asterisk 1.2 branch SVN 51363
zaptel svn 1980
libpri svn 393
addons svn 332
My
whether this
http://bugs.digium.com/view.php?id=8376
patch makes any difference. The problem is almost certainly not caused
by Centos (which is widely used with Asterisk) or EPIA (which I use
lots).
Regards,
Steve
On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote:
I have tried compiling asterisk
used in some of the
codecs...
Gordon
Regards,
Steve
On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote:
Hmm. Mantis says that in SVN 51223 it was implemented, im running
51363. However I may be wrong. I will apply that patch and let you
know.
Thanks for the pointer.
should I leave
using loopstart to connect the fxo to the avaya.
Some suggestions for busydetection?
Thanks,
--
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Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507
This is a G3. And I'm not the avaya operator. What do you mean with
2500 set and CPC?
On 1/29/07, C F [EMAIL PROTECTED] wrote:
What avaya system is this, if the avaya is configured on the ports to
use a 2500 set, then it should do CPC and should work as is.
On 1/29/07, Erick Perez [EMAIL
exten = _321[0123],n,AGI(agi://127.0.0.1:4577/call_log)
exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to)
exten = _321[0123],n,Voicemail,u${EXTEN}
exten = _321[0123],n,Hangup
comments?
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: unused variable `info'
make[1]: *** [codec_zap.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2/codecs'
make: *** [subdirs] Error 1
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Integradores de Telefonia IP y Soluciones Para
same with branch revision 53142
On 2/3/07, Erick Perez [EMAIL PROTECTED] wrote:
while compiling svn 53132 of asterisk branch 1.2
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i586
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on channel 'Zap/4-1'
but call can be processed normally.
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/1.4.x ?
Or do we have to tweak source code to balance loads (transcoding,etc)
between cores?
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improve my voice experience.
Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14?
thanks,
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Hi, I am looking to connect 66 analog phones to an asterisk box. I was
thinking of a Xorcom astribank 32port (2 of them and another 8 port).
this is because the phones have no near connection to an ip network,
so replacing the phones in favor of voip phones+network cabling is
kinda out of the
seems to have many names for the same thing...but
i better ask here and learn before i make a big mistake.
my customer has a dumb firewall (not SIP aware) that will not replace.
he wants another box to do the magic.
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Erick Perez
Cel
On Wed, Feb 11, 2009 at 1:56 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Wed, 11 Feb 2009, Erick Perez wrote:
Excuse my ignorance but if i have an asterisk in a LAN, and i have
users in their homes/internet (dozens), in order to correctly connect
those users across my firewall
(E1
signal can travel un-repeated over 5000 feet)
So far we are reading/evaluating about rhino channel banks and a quad E1/T1
(pci-e) on the asterisk box.
thanks again
--
Erick Perez
We are planning to run an outbound only campaign. A 20-second voice message
will be played to callers and our dialer on machine1 will send to
machine2-asterisk (1.4) instructions to dial 400 calls, play the message and
hang up. This will be done for about 1 million phones.
The asterisk box will
, 1 Apr 2009, Erick Perez wrote:
We are planning to run an outbound only campaign. A 20-second voice
message
will be played to callers and our dialer on machine1 will send to
machine2-asterisk (1.4) instructions to dial 400 calls, play the message
and
hang up. This will be done
.
Erick Perez
Cel +(507) 6675-5083
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I found this interesting but old white paper at Dell.com tech solutions and another one from INTEL.
It compares bandwidth usage of a PCI, PCI-X, PCI-E in33/66/100/133mhz bus and different technologies that can saturate the bus.
It helped me understand the bandwidth required for TDM
Hi,
I would like to read your comments for the following setup:
Building A:
3 voice E1incoming toa quad redfone fonebridge (TDMoE)
The fonebridge goes to a port in a 24 port gigabit switch
in the gigabit switch VLAN1 is for the fonebridge and the first gigabit NIC on a dual NIC server
in the
.
thanks in advance.
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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
gateway. Since he already have
some analog panasonic phones, he does not want to purchase Ip phones.
if you have some other ideas, let me know.
Ebay turned nothing in my searches.
Thanks,
--
Erick Perez
Panama Sistemas
Integradores de
?
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Erick Perez
Panama Sistemas
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Panama, Republica de Panama
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Hi,
Im doing some research for Disk on a Module (DOM)with asterisk realtime. To have no moving parts for a special project, I know I can use 3.5 or 2.5 HDDs but DOMs sound interesting.
Does someone have working experience with this?
Basically the Asterisk Realtime will be stored in MySQL and the
I understand Jeremy and Kris point of view (BTW Kris, astlinux rocks!!)
However the main question was not aswered (or i didn't get it, did I ?)
If I use a Disk on Module that has 2million hours MTBF and a Read/Write lifecycle of 2million times, then, How many days/weeks/months/years will take to
Jeremy, Cohen, Kris, thanks to all of you.
Indeed after reading the Sandisk paper it shed a lot of light on this matter. The whole idea is to have a large scale system with no moving parts (we call a large system something with250 users, at least down here ;-) )
the whole idea is for a customer
, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm just going to jump in here, and ask a stoopid question.
How could you possibly write a multi-user front end in AJAXwithout using a database backend like MySQL?
Doug.
-Original Message-From: Erick Perez [mailto:
[EMAIL PROTECTED]]Sent: Monday
This are the things that make me believe in technology. I wonder if Ubuntu Linux advocates will help with the development of the controlling modules.
*
Reuters 16:55 PM Oct, 11, 2006
AMSTERDAM -- Palm and pumpkin seed oil could soon be generating electricity to help power cell phone
://lists.digium.com/mailman/listinfo/asterisk-users
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Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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?
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is power.
On 10/18/06, Bob Chiodini [EMAIL PROTECTED] wrote:
On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote:
Hi people,
When you use a TDM400p with 4FXS i know i need to connect a 12V
connector to power the FXS lines.
Im not good at electric stuff so I ask...If I have a 60W DC to DC
adapter
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IRQ
problems.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Gustavo Alvarez
Sent: Thursday, May 12, 2005 2:29 PM
To: Erick Perez; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to decrease
Hi there, Being somehow new to this I like to be provided with
guidance as to how to diagnose a potential problem/bottleneck with my
GS ata286.
my internet speed is 512kbps downstream with 128 kbps upstream with a
local cablemodem provider. While i can make and receive calls
perfectly, my calls
Hi, we have set up a small project in a school the following way:
SITE_A(4 port analog to ip
g729)--ADSL_ISP1---ISP2Asterisk-PSTN
Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit)
The asterisk box gets internet service via a wireless antenna. 1 Mbit
of up/down
Thanks Rich, but i'm only allowed to use g729.
you said that some folks run high latency connections, but is 300ms
high in my setup?
On 3/19/06, Rich Adamson [EMAIL PROTECTED] wrote:
Erick Perez wrote:
Hi, we have set up a small project in a school the following way:
SITE_A(4 port analog
Hi, does anybody have a working config or tips to connect the welltech wellgate 3804 (4fxo) unit to asterisk via SIP ?I think I register it via SIP with my * box, but when sending calls from * to the wellgate the unit does not pass the call to any of the fxo ports.
thanks in advance,--
calls from the right of the diagram side to asterisk.On 3/23/06, Martin Joseph
[EMAIL PROTECTED] wrote:On Mar 22, 2006, at 10:24 PM, Erick Perez wrote:
Hi, does anybody have a working config or tips to connect the welltech wellgate 3804 (4fxo) unit to asterisk via SIP ? I think I register it via SIP
Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no hardware interfaces installed gives me this error. Im a bit new to this so any help will be appreciated. == Parsing '/etc/asterisk/musiconhold.conf': Found
Mar 26 00:58:49 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable
why should I? i thought in 2.6 kerneles that was not necesary when you dont have physical internfaces on the system.On 3/26/06, Jonathan Augenstine
[EMAIL PROTECTED] wrote:
Have you verified that ztdummy is loaded?On Sun, 2006-03-26 at 01:06 -0500, Erick Perez wrote: Hi, using asterisk 1.2.5
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