I have created the following stuff:
BRANDS:
Brand Name Language Published Number DID Inc Markup (in 1/100 of 1%)
Gamma en 5000 6 0
TRUNKS:
Trunk Name Technology Peer/Trunk
Gamma SIP 200.68.XXX.XXX
ROUTES:
Pattern Comment Trunks Connect Fee Inc. Seconds Cost per additional minute
63
Kido,
I start to compile openh323 1.3.15 as recomend in the
asterisk-oh323-7.0 README but I get an error:
h323ep.cxx: In member function `virtual BOOL H323EndPoint::IsLocalAddress(const
PIPSocket::Address) const':
h323ep.cxx:2397: no matching function for call to `PIPSocket::Address::
2004 20:41:17 -0300, Nahuel Alejandro Ramos
[EMAIL PROTECTED] wrote:
I have already tryed this but asterisk always send root as h323_id
On Sun, 28 Nov 2004 21:31:52 -, kido noagbodji [EMAIL PROTECTED] wrote:
If you are using GnuGK, i think this should do,
in your h323.conf file
Thank you very much. I have been trying it but I get in trouble installing it
I will try it again.
Nahuel Ramos.
On Fri, 26 Nov 2004 12:11:20 +0200, Michael Manousos
[EMAIL PROTECTED] wrote:
Thanks.
I appreciate that.
Michael.
kido noagbodji wrote:
i have had some
I use ASTCC and works perfect for Prepaid situations.
Nhauel Ramos.
On Mon, 29 Nov 2004 16:48:46 -0400, Sebastian Atala [EMAIL PROTECTED] wrote:
Is anyone successfully using asterisk-prepaid-0.3.1?
I try to configure but doesn't work. It said that you need to do a few step,
copy a
Hi,
Anyone know how can I send a username or account id (h.323) and a
password to register on a remote Gatekeeper. I am using the Nuphone
channel with the h323.conf. I tryed everything but Asterisk always
send root as account id and the Gatekeeper rejected me.
Thank you very much...
Hi,
Anyone know how can I send a username or account id (h.323) and a
password to register on a remote Gatekeeper. I am using the Nuphone
channel with the h323.conf. I tryed everything but Asterisk always
send root as account id and the Gatekeeper rejected me.
Thank you very much...
Hi,
Anyone know how can I send a username or account id (h.323) and a
password to register on a remote Gatekeeper. I am using the Nuphone
channel with the h323.conf. I tryed everything but Asterisk always
send root as account id and the Gatekeeper rejected me.
Thank you very much...
] Username
type=h323
e164=99
context=test
K.
- Original Message -
From: Nahuel Alejandro Ramos [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, November 28, 2004 5:52 PM
Subject
endpoint is registered and if you modify you
extensions.conf file like this it should work
exten = 99,1,Dial(h323/12345678)
assuming that your h323 registered endpoints IPN(ANI) is 12345678.
That should work.
K.
- Original Message -
From: Nahuel Alejandro Ramos [EMAIL
Hi everyone,
Could someone help me on make my Asterisk registers to a Gatekeeper.
I have compiled the chan_h323.so and it seems to be working.
What I want to know is how can I route my SIP clients to a single
account on a remote Gatekeeper.
I have tried a lot of conbinations but nothing
Hi everyone,
Could someone tell me if I could make a Asterisk PBX + Digium
hardware with more than 20 FXS? If I have a 5 PCI PC, I could only
plug 5 TDM40B (4-port FXS) maximun. Is there a solution, a hardware in
Digium or something that let me have a PBX with about 50 or 100 FXS
and internal
voice channels theoretical max -- and go to town. :)
http://www.digium.com/index.php?menu=hardware_products
All of that hardware works splendidly with Asterisk. You can peruse the
Wiki for more information on configuring the beasts:
http://www.voip-info.org
Greg
Nahuel Alejandro
-0300, Nahuel Alejandro Ramos said:
Thanks for the quick answer. I have seen this cards but how do I
convert an E1 to 30 FXS plugs? Sorry about my newbie question about
hardware. I have got working an Asterisk with Xlite and ATAs, but I
need a lot of internals phones.
Thank you again
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nahuel
Alejandro Ramos
Sent: Monday, November 15, 2004 3:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] $10 for G.729 ?
Hi,
I have
Kanuri
-Original Message-
From: Nahuel Alejandro Ramos [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 16, 2004 3:11 PM
To: Kanuri, Seshu (Company IT)
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] $10 for G.729 ?
Seshu,
Yes, I have
Hi,
I have already send a message to Digium but I post this again to
know of anyone who use the G.729 codec.
My questions are:
- Do I have to pay 10$ per month or it is only this fee to use the
codec forever
- How many licenses do I have to buy ? ( I have an Asterisk talking
with two Cisco
Hi everyone,
I have my asterisk working with a Cisco 2610 PSTN Gateway connected
over SIP protocol. Could anybody tell me if I can send and receive SMS
through this Gateway with the SMS command in asterisk?
I have been searching for on google and I saw that Asterisk can send
SMS over IAX to
You have to use DeadAGI instead of AGI to call your script, for example:
exten = 77,1,Answer
exten = 77,2,DeadAGI(astcc.agi)
exten = 77,3,Hangup
Regards..
Nahuel Ramos.
On Thu, 4 Nov 2004 13:14:56 -0400, Victor Cartes
[EMAIL PROTECTED] wrote:
Does anybody know how to stop the AGI
Hi,
I am living a wear thing. I am using my asterisk with all kind of
NAT / PAT / NPAT, with multiple ports on the same network address and
all works perfect.
The problem I have been trying to solve is with a Cisco ATA behind a
Linksys NAT. The ATA's register work ok, but when I execute sip
] wrote:
Mark Phillips wrote:
Do you have nat=yes in your relevant sip.conf stanza?
I would look for an option that is Block WAN Requests Set this to
disable, then your linksys will respond to pings. Hope this helps.
JB
- Original Message -
From: Nahuel Alejandro Ramos
(Company IT)
Sent: Tuesday, October 26, 2004 2:49 PM
To: 'Nahuel Alejandro Ramos'
Subject: RE: [Asterisk-Users] ASTCC with password
Thanks Nahuel. I will try this tonight.
Seshu
-Original Message-
From: Nahuel Alejandro Ramos [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 26
Hi,
Someone know where can I buy Digium Hardware near of Argentine,
Brazil or Chile. I am from Argentine.
Thank you very much...
Nahuel Ramos.
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[EMAIL PROTECTED]
I have read all the wiki, compile and install Asterisk with MYSQL_PEER
MYSQL_FRIENDS equal 1. Then I created the tables and test the
retrieve_extensions_from_mysql.pl files and it add me a correct .conf
file.
My question is, there were a configuration for asterisk to read
directly the
)
Thanks for your help.
You are welcome.
Seshu Kanuri
Nahuel Ramos.
-Original Message-
From: Nahuel Alejandro Ramos [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 02, 2004 9:20 AM
To: Kanuri, Seshu (Company IT)
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Mark,
I downloaded the Winamp plugin and it works great.
Thanks...
Nahuel Ramos.
On Thu, 28 Oct 2004 00:32:53 +0200, Carl Sempla [EMAIL PROTECTED] wrote:
On Wednesday, 27 October, 2004 20:00 : Mark Halverson [EMAIL PROTECTED] wrote:
I play em just fine using Media Player
Yo estoy armando un Asterisk pero el PSTN Gateway es un CIsco 2610.
Has tenido buenos resultados con la E1 directamente en el Linux? Te
demande mucho hardware?
Gracias...
Nahuel Ramos.
On Tue, 2 Nov 2004 13:31:37 -0300, Horacio J. Peña
[EMAIL PROTECTED] wrote:
Gracias por el dato. Una
Hi everyne,
I need some help configuring SIP Friends on MySQL. I have already
compile and install Asterisk with the MYSQL_FRIENDS = 1 and created
the MySQL databases and tables, but I do not know how to call the
tables from my sip.conf.
Please, anybody could help me telling me what file I have
Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: Darren Wiebe [EMAIL PROTECTED]
To: Nahuel Alejandro Ramos [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 10:42 PM
Subject: Re
Hi everyone,
I searching for a plugin for ASTCC to add a simple password enter.
So if a person has a valid calling card number must enter a secret
password to make a call.
Thank you very much...
Nahuel Ramos.
___
Asterisk-Users mailing list
this but not
with much success.
I need the ASTCC part also, including the perl scripts and config files,
in addition to the Asterisk Extensions file
Thanks in advance
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nahuel
Alejandro Ramos
Sent
Brian and Kristian,
I have the same trouble, please could you let me know which is the
new link for www.bkw.org/load.txt because I can not enter. Do I have
a privileges to enter?
Thank you very much...
Nahuel Ramos.
On Fri, 22 Oct 2004 17:49:38 -0500, Kristian Kielhofner [EMAIL
I do not know how Bicon works. You could send a mail to their support section.
A question: could you add exten = .. to your ext2.conf?
On Tue, 26 Oct 2004 13:54:22 -0400, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
Nahuel,
I have the ASTCC installed. My problem is that I don't
this tonight.
Seshu
-Original Message-
From: Nahuel Alejandro Ramos [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 2:45 PM
To: Kanuri, Seshu (Company IT)
Subject: Re: [Asterisk-Users] ASTCC with password
As I can see, you have to put under the [defualt] contexto
Hi everyone,
Please, anyone can help me calling my extensions from a mysql. I
have created the DB, compiled my Asterisk with MYSQL_FRIENDS and all
attributes with 1, but I do not know how to call my DB from
extension.conf or which file a should modified. I need to use
extension.conf and sip.conf
Hi everyone,
I am using astcc for my prepaid calling card and the billing goes
perfect, but when I make a call I do not hear any voice. The strange
thing is that when I press a number on my Xten, on my ATA I hear a
simple beep for all the numbers. I have tryed to change the codecs
but I still
was my fault!!
I have repaired my error. The problem was on my sip.conf i would not
add reinvite=yes and canreinvite=yes.
Nahuel Ramos.
On Tue, 26 Oct 2004 18:14:57 -0300, Nahuel Alejandro Ramos
[EMAIL PROTECTED] wrote:
Hi everyone,
I am using astcc for my prepaid calling card
Hi,
Anyone could use Asterisk Prepaid with a MySQL database? Thanks.
Nahuel Ramos.
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To UNSUBSCRIBE or update options visit:
Can I post this again? :)
Where do I register the SIP clients for more performance? (Asterisk or SER)
Thank you very much.
Nahuel Ramos.
On Fri, 22 Oct 2004 02:53:36 -0700 (PDT), Asterisk .
[EMAIL PROTECTED] wrote:
Hi,
--- Iqbal [EMAIL PROTECTED] wrote:
asterisk would only be
Performace: CPU MEM per Call.
I will use SER to route call but I want to know if it is better to
register SIP clients on SER or Asterisk?
Thanks.
On Fri, 22 Oct 2004 10:24:17 -0700 (PDT), Asterisk .
[EMAIL PROTECTED] wrote:
--- Nahuel Alejandro Ramos [EMAIL PROTECTED] wrote:
Can I post
To helpful link!. It answer lot of my question.
Thank you very much Asterisk in Yahoo...
Nahuel Ramos.
On Fri, 22 Oct 2004 11:46:46 -0700 (PDT), Asterisk .
[EMAIL PROTECTED] wrote:
--- Nahuel Alejandro Ramos [EMAIL PROTECTED] wrote:
Performace: CPU MEM per Call.
I will use SER
Hi everyone,
I am looking for information about performance on Astrrisk especialy
using astcc.
Could anyone send me a table or something like that with columns CPU
usage, memory usage and calls supported with this platform.
How many calls can it support simultaneously? How many calls per
second
Hi everyone,
I have some doubt about use or not to use SER.
I need a solution using a single linux box that manages, aproximatly
500-1000 registred SIP users, but not more than 50 simultaneouly
calls. My plan is to use between 3 to 5 Cisco 26xx PSTN gateway in
diferents cities of my country
Hi everyone,
I am looking for a prepaid billing solution for my VoIP. I have
already install SER but it does not support a statefull tracking of
the call, so it is dificult to hang up a call when it has zero credit.
Posting messages on Serusers maillist there are a lot of people
telling me to
Thanks Senad, but I am looking for free software.
Nahuel Ramos.
On Wed, 20 Oct 2004 18:49:34 +0100, Senad Jordanovic [EMAIL PROTECTED] wrote:
Hi,
Have a look at this for your needs...
www.bicomsystems.com
Regards,
Senad J
Nahuel Alejandro Ramos wrote:
Hi everyone
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