server or it has happened a couple of times when I
redirect my desk phone to my cell.
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Tel: +52-55-91169161 Ext 2001
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On Fri, 1 Apr 2005 14:50:45 -0700, Wiley Siler wrote
Un what is todays date?
You will have to excuse the people who live outside the USA that do not
know that April 1st is April Fools day as it is not an international thing.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones
.
The last time I had this problem the solution was to use Function 808 on
the Nortel phones to enable long tones. But this is not working here. Was
there a change in Asterisk that affected this? Has anyone found another
solution?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones
problems.
-Scott
You have to activate IAX support by hand inside your FWD account. Go into their webpage and find the option to activate IAX, after you select it it should take about half an hour until you can use IAX.
--
Carlos Chavez
Director de Tecnología
Does anyone have a document on how to implement R2 for use in Mexico?
What packages do I have to download and compile?
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Telecomunicaciones Abiertas de México S.A. de C.V.
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Core 1 but I plan to install new servers
using Fedora Core 3
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to take care of is that every card gets its own
IRQ and does not share it with anything else.
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http
I want to buy a new server to run Asterisk and after looking at prices
for the Athlon XP 3000+ it costs the same as an Athlon 64 at the same speed
rating. I was wondering if Zaptel/Asterisk will compile/work on an Athlon 64?
--
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Director de Tecnología
Telecomunicaciones
run, but I
was wondering if I had to use a 32 bit distribution an not a 64 bit one. I
use Fedora Core 3 on my current server which has an Athlon XP 2100+. It is
good to know that I can use the native 64 bit version of FC3 so the rest of
the system will be optimized for the new processor.
--
Carlos
by themselves.
Any idea?
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see if the init.d script you are using to start Asterisk is
changing the default font set from ISO-8859-1 to -5 or something else. Maybe
when you installed Fedora you checked that you wanted support for Russian.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de
. Externip should be the external public IP.
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Telecomunicaciones Abiertas de México S.A. de C.V.
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-lang)
exten = 1,1,Goto(spanish,s,1)
exten = 2,1,Goto(english,s,1)
[spanish]
exten = s,1,SetLanguage(es)
... do something else
[english]
exten = s,1,SetLanguage(en)
.. do something else
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext
On Thu, 2005-03-03 at 17:43 +0500, Rizwan Chaudhry wrote:
I have linux 2.6.5 running on my machine.I downloaded The latest
version of Zaptel from the cvs repoistory.Compiled zaptel with the
make linux26 option. Installed it by modprobe which gave no
errors.However when i did modprobe wctdm i
) but appartently
the DTMF tones are not passed to asterisk and the call cannot proceed.
This only happens when calling from a digital phone on the Nortel. If I
connect an analog phone to the PBX and dial from there the call can go
through. Anyone has experience with this?
--
Carlos Chavez
Yes the norstar is a piece of work. We have the same problem. The
only solution is to dial asterisk then press feature 808 on your
digital phone then the phone system will pass the dtmf - what a crock!
How do you press feature 808 on the phone?
--
Carlos Chavez
Director de
if it
is possible to change the DTMF duration in Asterisk to a maximum of 120ms?
They say that their system only sends tones of that length and that is why
Asterisk does not detect the tones. Can this parameter be manipulated on
Asterisk? Is it Asterisk or Zaptel who is responsible for this?
--
Carlos Chavez
= 255.255.255.0 ; Internal netmask
context = (my context) ; Default for incoming calls
[4002]
type=friend
username=4002
secret=secret
host=dynamic
amaflags=default
accountcode=temp
callerid=Carlos Chavez 4002
nat=yes
mailbox=4002
The other extensions I am trying to call
to do.
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On Wed, 18 Feb 2004 10:26:07 +0200, Dan wrote
Hi,
Quoting Carlos Chavez [EMAIL PROTECTED]:
I am having trouble trying to get DIAX to answer calls. Ir
registers
with my * server and I can make outgoing calls without any problems. If
I
dial a DIAX client from any other
Unmonitored
I tried the phone both on the local network and from another network.
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like the
quality, but not many hard phones support it. What is the most common low
bandwidth codec? If I buy phones from several providers, what codec would you
recommend?
--
Carlos Chavez
Computer Engineer, CCNA
Corporativo Lacer S.A. de C.V
Is there a way to know which Codec a particular phone is using? I have
several devices which support different codecs and I would like to find out
which one was negotiaded with Asterisk. Is there a CLI or Manager command to
get this information?
--
Carlos Chavez
Computer Engineer, CCNA
Yesterday I tried to connect an * server with an X100P card to an
extension of an ATT PBX. The X100P never could detect the line and always
gave an alarm. Is there some special type of config that must be done to
connect an FXO port to an extension of a PBX?
--
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Corporativo
not hang up before 30 seconds, my machine then slows down and it
can take up to 10 minutes to shut down. Is speex worth the trouble?
I am using Asterisk CVS-03/17/04-20:59:14 on Fedora Core 1 with Speex
version 1.0.3.
--
Carlos Chavez
Computer Engineer, CCNA
Corporativo Lacer S.A. de C.V
faxdetect=both on my /etc/asterisk/zapata.conf file. Could this
be the problem? Could it be that my asterisk server detects the transfer as a
fax signal?
--
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Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V
I see that I can purchase G.729 licenses for my Asterisk server, but I
have seen that many phones support a G.729 variant like A or B. Are these
suppoted by the same G.729 codec in Asterisk?
--
Carlos Chavez
Computer Engineer, CCNA
Corporativo Lacer S.A. de C.V
anyone
have any recommedation?
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Computer Engineer, CCNA
Corporativo Lacer S.A. de C.V.
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.
When talking to Telmex you have to ask for their Video Conference service
which is where they use ISDN instead of R2.
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Corporativo Lacer S.A. de C.V.
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the stock kernel and you should be fine. I have 6 servers using FC1 and they all compile without problems.
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Computer Engineer, CCNA
Telecomunicaciones Abiertas de México S.A. de
C.V.
when
concatenated to other numbers. When you have to say 101 the 100 sound changes
to: ciento uno.
Thank you for your help.
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Computer Engineer, CCNA
Telecomunicaciones Abiertas de México S.A. de C.V.
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digits. Since the upgrade
all sounds play as en.
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On Thu, 06 May 2004 18:45:15 +0100, Fran Boon wrote
Carlos Chavez wrote:
ok, we have an 'es' syntax for saynumber() but it doesn't seem to
support ciento uno as yet.
Is this the only number that changes?
What about 102? 110? 1001?
All numbers like 10,20,30,40,50,60,70,80,90 and 100
module without the computer crashing. I have already erase the entry in
/etc/sysconfig/hwconf and turned kudzu off during boot.
Anyone know of a way to fix this (short or reinstalling FC2)?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V
On Sun, 2004-10-24 at 05:10, Tzafrir Cohen wrote:
One obvious solution is not to automatically load kudzu.
chkconfig --remove kudzu
Another obvious solution of the same sort is modprobing the zaptel
module earlier in the boot process.
I can't seem to figure out , though, where kudzu
On Fri, 2004-10-29 at 12:17, Matt Schulte wrote:
I am having the same problem, it doesn't work on my SNOM either. Below
is my sip.conf .. On both sipura and SNOM I get same results, I can hear
voice
but not send voice.
When you do a show g729 on the CLI do you get that the license is
site where the firmware is. New firmware is at version 1.0.0.41
There is still a very big problem with this phone, the dial plan will
only allow you to dial 10 digits. For local numbers this is not a
problem, but you cannot dial long distance.
Carlos Chavez
between both version.
Ok, I did the change you specified and now we can receive calls from Nextel
phones but get no callerid on any call. How do I apply the patch to libmfcr2?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext
I have a customer that has recently upgraded their network and now
their Aastra 9133i phones are loosing their connection to the Asterisk
server. They were using an external Asterisk server and now we have
installed a new internal server with Asterisk 1.4.8 on a SIP/IAX
implementation
between both version.
I patched mfcr2.c but I still cannot receive calls from Nextel phones
unless I put ANI to 0 on unicall.conf
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
On Mon, 2007-07-23 at 11:47 -0500, Moises Silva wrote:
Alvaro,
Naming Asterisk versions is of little help since Asterisk is not
the one failing here. It would be more helpful know the libmfcr2 and
spandsp versions that were used in the working/non working tests, is
that possible? do you
On Fri, 2007-07-27 at 11:09 -0500, Victor Toofic wrote:
Hi,
zaptel.conf:
span=1,0,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
loadzone=mx
defaultzone=mx
unicall.conf
[channels]
context=incoming
usecallerid=yes
hidecallerid=no
It seems the problem with Unicall and Nextel is also present in
Asterisk 1.2 and not only in 1.4. I decided to downgrade from 1.4.9 to
1.2.23 so the customer could have CID and calls from Nextel but today he
told me that they cannot receive any calls from Nextel, they get a busy
tone
Here is a log with level 255 when a Nextel phone tries to call in:
Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ]
Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1
On Fri, 2007-08-03 at 00:23 -0300, Luis Antonio Prata Barbosa wrote:
Hi Carlos,
I suggest you download spandsp-0.0.3pre22.
(http://www.neuwald.biz/files/spandsp-0.0.3pre22.gz)
I don´t know why , spandsp after that uses digits 1,2..8,9,A,B,C,D,E,F
instead of 1,2,..,9,0,A,B,C,D,E. So,
On Wed, 2007-08-08 at 17:08 +, John Meksavan wrote:
Wes,
What kind of service outages did you experienced?
This would use for my office and I cannot afford for any dropped calls or
poor audio quality, when talking to customers.
My experience with Voicepulse has been good
I am having a bit of a problem implementing the pickup command in my
dial plan. I have setup this rule:
exten = _*8XXX,1,Pickup(${EXTEN:2})
This works as expected when someone dials an extensions number and I
can get the call. The problem I have is that when a call enters my
I usually have good results when using a regular fax machine connected
to a PAP2T on a small network. I have a customer that has this setup in
several offices. Lately I have noticed that recent versions of Asterisk
have worse results with this fax setup that onlder versions. I have 3
On Mon, 2007-08-13 at 15:25 +0530, [EMAIL PROTECTED]
wrote:
Hi,
I have successfully configured DIGIUM card and successfully communicated
through it to the another E1 card running application. Can anybody tell me
does TE120P
support MFC/R2 protocol.
Thanks and Regards
sanchal singh
Here is Mexico the phone company uses a DSL router from 2Wire which in
my opinion is quite bad. I am having problems getting PAP2T adapters
connected to Asterisk using these routers. They connect fine but after
about 5 minutes I get a message on the Asterisk console that the ATA is
On Thu, 2007-08-16 at 16:23 +, John Meksavan wrote:
Asterisk Users,
I have 3 FXO modules with the TDM400P Digium Card. I can dial into the
Asterisk rings my Sip phone, but dialing out with my SPA941 phone through
the zap channel is a problem. I keep getting this message on the
On Mon, 2007-08-20 at 11:45 -0500, Jordan Novak wrote:
I am trying to figure out how long a caller waited in queue for
someone to answer versus how long they stayed on the phone after the
answer. I am assuming that the duration is the total talk time and
that the billsecs are the total time in
On Wed, 2007-08-29 at 00:03 +0300, Tzafrir Cohen wrote:
On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote:
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz
with Asterisk 1.2.22 somewhat successfully.
Shouldn't you
I was wondering if anyone has an easy way to emulate dialing in a round
robin fashion like when you use Zap/r1 for Zap trunks. At the moment
what I do is simply make a macro that will dial the sip trunks in order
so if the first one fails it goes to the second and so on. The problem
with
I am having a strange problem with an Asterisk server that has a small
5 seat call center. While everything seems to be working properly I if
do a core show channels the server goes into a loop:
pbxinsol*CLI core show channels
Channel Location State
I have a customer that has two buildings that are connected with a
fiber link. We have a single Asterisk server to cover both buildings.
Now the customer went and bought an overhead paging system for the
remote building and they want to integrate it with Asterisk. Is there a
device that
On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote:
Dear All,
I'm integrating avaya commuication manager difinity ver 1.0 with
asterisk using B2B E1. following are the details of my H/W,
zaptel configs and software installed.
Digium TE110p
asterisk 1.2.19
cent OS 4.4
zaptel
I have several installations of Asterisk (several versions) where we
have our own web interface that uses Mysql and Realtime. When we do
modifications to Mysql we use a Manager connection in order to reload
the configuration (we use Realtime static for extensions) sometimes
Asterisk will
On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote:
Hi folks:
I know it's come up a few times before, but I need some more detail.
I'm looking for a SIP DECT (cordless) phone for North American
installations. I've heard only of the Siemens Gigaset S450/C450 phones.
Apparently these
Does anyone know if the Dell Power Edge 1900 has an issue with
multiport E1 cards? We've had this server running for a while now with
2 E1 cards. At first we tried to install an Openvox D210P card with two
E1 ports but after a couple of kernel panics we thought that maybe the
card was
when you click a
button. That will fire an event that connects to the manager interface and
uses originate to dial the external call and then dial the internal extension
if all conditions are met. The numbers will be in a database.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones
On Fri, 2007-09-21 at 14:03 -0500, Ricardo Melendez wrote:
Help I need to install asterisk 1.4.X using unicall, somebody can tell
me which are the correct versions of spandsp, libunicall, libmfcr2,
libsupertone, to install with asterisk 1.4, I have installed a
prepatched version, but I need to
On Tue, 2007-09-25 at 10:59 +0200, Erik Wartusch wrote:
Hi,
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a
business graded installation (with really traffic on not 3 calls a
day ;-) )
Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall)
On Thu, 2007-10-11 at 15:07 +0200, Vincent wrote:
Hello
Has someone used the OpenVox A400P01 (ie. a supposedly
Digium-compatible A400P board with a single FXO module
www.openvox.com.cn/products_detail.php?genre_id=9id=28) successfully?
I've put it in an older PC with a Gigabyte GA-7ZX
I have a customer that recently started having a problem with their
Call Center SIP extensions. The problem is that after some time the
caller will hear a triple tone (beep, beep, beep), a 5 second pause,
another triple tone and then the call will be dropped. This usually
happens between
Is there an example on how to use two E1 ports connected to each other
to simulate connections? Since I do not have an E1 at the office I need
for one port to act normally and the other to act as if it were the
telephone company so I can send calls from one E1 to the other. Someone
has
I am having a bit of a problem getting AMD to work on a new server. On
my regular office server it works like a charm. I am running Asterisk
1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and
I am using a SIP trunk to send out calls (the same one on both servers).
We have a customer that has Asterisk 1.4.12.1, Zaptel 1.4.5.1,
Asterisk-Addons 1.4.3. running on a Dell Poweredge 1900 server (Dual
Core Xeon, 4gb RAM, 500gb Raid 5). Until a month ago they had two
TE120P cards and everything was working fine. Since they needed to add
a third E1 line we
On Wed, 2007-10-17 at 21:03 +0300, Atis Lezdins wrote:
On Wednesday 17 October 2007 19:09:23 Carlos Chavez wrote:
Why would inserting a multiport card affect Asterisk and the
server? How can I debug this situation? I do not have enough slots to
insert three single cards of the same
I have a customer that needs an Asterisk server to sell minutes for
cell phones in Mexico. I do not see a problem with that since he will
get the calls by SIP and then use GSM adapters to get the calls into the
GSM network. My problem is that his customers only want to be
identified by
On Mon, 2007-10-22 at 15:35 -0400, Rurouni Alucard wrote:
Saludos Carlos,
Como vas a recibir las llamadas via SIP, puedes especificar el IP del
host que te enviara las llamadas, por ej.
Este es un bloque que tengo definido en el SIP.conf de uno de mis
servers para enrutar las llamadas
On Mon, 2007-10-22 at 15:13 -0400, [EMAIL PROTECTED] wrote:
On 10/22/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I have a customer that needs an Asterisk server to sell minutes for
cell phones in Mexico. I do not see a problem with that since he will
get the calls by SIP
On Fri, 2007-10-26 at 16:35 -0400, Michelle Dupuis wrote:
I have a new asterisk system with a T1 card. It appears that running
ztcfg -vv is required in order for asterisk to start properly.
Is this correct? Are people adding this command to the asterisk
startup script?
This
immediately. Obviously this creates problems when I am dialing long distance
numbers or anything that needs more than 8 digits.
Is there any way to increase the number of digits before the number is
diales automatically?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de
.
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Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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should change the DEFAULT_T1 value of mfcr2.c fomr 5000 to something like 2. I also included a bit timeout of 120 seconds in the dial command. For the moment every call is going through although I still have some testing to do.
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Director de Tecnologa
Telecomunicaciones
Does anyone know if it is possible to upload a common directory to all
Aastra phones (480i, 9133)? Is there someting equivalent to the way Polycom
phones do it?
--
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Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
ATA
devices in the past to connect analog faxes and they usually do not have any
problems. Any sugestions?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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Tel: +52-55-91169161 Ext 2001
signature.asc
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the DID on
the E1 for faxes so I need to fix this.
The fax is connected to a Linksys PAP2 adapter but I have also tried
rxfax and I get the same results when trying to use the E1 connection. Is
there a setting or modification that can be done to unicall?
--
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Director de Tecnología
Does Unicall support disabling echo cancellation on an E1 circuit when a fax tone is detected? I think this is the reason why I cannot send or receive faxes on my Asterisk server.
--
Carlos Chavez
Director de Tecnologa
Telecomunicaciones Abiertas de Mxico S.A. de C.V.
Tel: +52-55
I do not know if this will make a difference but the protocol-variant
for Mexico should be:
protocol-variant mx,10,4
You only get 10 digits from the phone company.
On Wed, 2008-02-27 at 18:03 -0800, Andres Tello Abrego wrote:
protocol-variant mx,20,4
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I just connected an Asterisk server to an Avaya Pbx using the
instructions at: http://www.voip-info.org/wiki/index.php?page=Asterisk
+Avaya
Everything seems to be working as I can send and receive calls. The
only detail I am having a problem with is that when an extension on the
I have a new installation where an Asterisk server is connected to an
Avaya PBX via a PRI E1. We are having a problem that I attribute to
their firewall but I just want to make sure.
When we make a call from the Avaya to a SIP extension there is only
sound on the receiving end.
I have a customer with a Fortinet Firewall that is having stability
issues with Asterisk and SIP endpoints (PAP2T) outside his network.
The first issue I see is that Asterisk sees all phones as the IP
address of the Fortinet. Since the parameter localnet defines the
local
I need a refresher course on how many licenses I need to buy. I have
an Asterisk server that receives calls by SIP (G729) and then sends them
to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if
the license is per channel or per call so I do not know if I need 32 or
64
I have a big headache. I have an Asterisk server connected to an Avaya
PBX. Everything is working between those two. The problem is that I
have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the
Internet to the Asterisk server through a Fortinet firewall. When
calling from a
I am still having a very frustrating problem win an Avaya-Asterisk
system. I have written about this before but I am expanding the
description of the problem just in case someone can give me some
insight.
This installation is an Asterisk 1.4.19.1 server connected to an Avaya
PBX
parameters correctly.
2) Also in sip.conf, try the following on the PAP2's sections:
disallow=all
allow=alaw:10
In case that fails, try also
disallow=all
allow=alaw:20
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Carlos Chavez [EMAIL PROTECTED
Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Carlos Chavez [EMAIL PROTECTED] escreveu:
I am still having a very frustrating problem win an Avaya-Asterisk
system. I have written about this before but I am expanding the
description of the problem just in case someone
I have an Asterisk 1.4.19.1 server that is behind a Fortinet firewall.
Localnet is 192.168.2.0/255.255.255.0 and all external sip devices look
as if they are on the same local network because the Fortinet rewrites
the incoming IP as its own address.
The problem I have is that when
I am a bit desperate trying to solve this problem. Sorry if I am
abusing the list a bit with the same king of question.
The problem I am having is very specific which is why it is very
difficult to diagnose and fix. Basically an Asterisk server is
connected via E1 PRI to an
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would like
to busy out that line. Since that line is the head of the hunt group I
cannot simply disable that channel, I need to busy the line so calls
will
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Canall Tecnologia em Comunicações Ltda.
- Carlos Chavez [EMAIL PROTECTED] escreveu:
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would
like
to busy out that line. Since
Thank you. Unfortunately the phone Company in Mexico is not very
helpful when it comes to those services.
On Tue, 2008-05-20 at 16:48 -0500, Tilghman Lesher wrote:
On Tuesday 20 May 2008 16:08:19 Carlos Chavez wrote:
The problem is that I do not have physical access
Don't know about Debian but in Fedora or CentOS you need to install
mysql-devel to compile Mysql support in Asterisk-Addons
On Wed, 2008-05-21 at 14:31 -0500, JR Richardson wrote:
Hi All,
I'm poking around with 1.6, tried to compile the addon package, but it
doesn't see mysql_config
As long as each tenant has its own context you can use the same
numbering plan. The only thing you need to keep unique are the names
for the SIP devices. If you want your tenants to be able to call each
other then you would need to set up a special prefix for each tenant.
On Thu,
?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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Since Zaptel 1.4.11 has been released, why is the link on the Asterisk
website pointing to 1.4.10.1? Is there a problem with the newest
version or just someone forgot to update the link?
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Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
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