Is there an up and running provider of SIP termination in Brazil?
I know that there are some people building on a SIP
termination solution.
But who as it up and running ?
Best regards,
Han
___
Asterisk-Users mailing list
[EMAIL
Has anybody any idea what I can do with asterisk following the
Brazilian law.
I do not have a multimedia license or a telecom license, but
I ace asterisk.
Are there companies who are looking for asterisk expertise
in Rio de Janeiro?
Greeting Han
__
For so far as I know. Asterisk is an open source and allowed with all the
hardware it will make the product even stronger.
For so far as digium they have a great marketing product in hands.
Everybody knows Digium as a great supporter for asterisk and a 100%
compatible hardware supplier.
Digium, pl
Well so far as I know there is no preferred
version of Linux for Asterisk.
It will work with the most versions of
linux.
For myself I have it perfectly working
with new hardware on a fedora core 2 version with a 2.6 kernel
But be sure you can not follow the
asterisk installation by th
Is there a solution for asterisk to send the calling costs
to a display of a grandstream Bt101 phone.
Does anybody know if there is a solution for this?
Greetings Han
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.dig
Mike,
Sometime ago there was a message from rits they can help you out.
I think it is one of the biggest Voip companies in Holland
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas Sikkema
Sent: woensdag 18 augustus 2004 6:17
To: [EMAIL PROTECTED]
Subject: [OT] RE: [Aster
For asterisk I am using more than one sip providers.
The provider in Holland would like to have the international calls like
00 31 20 1234567 but the provider in the US likes it like 0011 31 20 1234567
Can I make a rule in asterisk so that I can dail 00 31 20 1234567 and
asterisk dails 0011 31 20
Johannes van Hulst
Sent: Thursday, August 26, 2004
9:53 AM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] need
help with zaptel.
Edward,
I had the same problem I am running fedora
core 2 with a 2.6.8-1.521 kernel at it is working now perfect.
Craig wrote the following :
I have had
Edward,
I had the same problem I am running fedora
core 2 with a 2.6.8-1.521 kernel at it is working now perfect.
Craig wrote the following :
I have had success with this
using both the X100p (wcfxs and wcfxo) and TE410p (wct4xxp) under Redhat FC2
2.6.5. The instructions are on t
hardware or OS problemmake[1]: *** [app_voicemail.o] Error 1make[1]: Leaving directory `/usr/src/asterisk/apps'make: *** [subdirs] Error 1US040814:/usr/src/asterisk #
Who has the same experience with suse
linux or who can help me with resolving this problem
From:
[EMAIL PROTECTE
As usual,
however, YMMV. Let me know if this works for you.
Sincerely,
Trevor Hammonds
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Johannes van Hulst
Sent: Monday, August 16, 2004 9:30
AM
To:
[EMAIL PROTECTED]
Subject: [Asteri
Have anybody experience with the following error on a linux system.
The system is for the rest running perfect without problems.
The system was full installed with Suse 9.1 and updated.
Following uname is my kernel 2.6.5-7.104-default
Greatings Han
Suse 9.1 professional
AM
How has experience in Asterisk voip provider?
I am trying to setup a reliable Linux system with Asterisk
for a voip provider.
Therefore I got two more or like identical systems.
System 1
AMD Atlhon XP 2200
Asus A7V600-X bios 1002
1Gb memory 333 Mhz
Asus 7100 videocard
120GB ha
I have a hard time to forward all my outgoing calls to my SIP
connection.
For incoming calls I set up the registry and it is working perfectly
How can I tell asterisk to forward all calls beginning with 9
to
Gate.Sipserver.com
User : userxx
Password : pwxx
Best regards,
Ha
Hello,
Who can help me I am trying to setup the sendmail so that I can
mail the voicemail’s to an internet SMTP mail server.
I know that I have to setup the sendmail.cf and configured a
relay to my normal SMTP server.
I am running RedHat 9 and my internet provider has a SMTP mail
s
Is it possible to configure a couple of SIP accounts for incoming
calls and then forward this calls to different SIP phones?
Like
Registry [EMAIL PROTECTED]
Registry [EMAIL PROTECTED]
Forward 123 to [EMAIL PROTECTED]
Forward 440 to [EMAIL PROTECTED]
Regards,
Han
Gustavo,
Als je duits nog een beetje goed is heb ik hier een duitse provider voor ons
MVG Han
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Holger Schurig
Sent: Thursday, July 15, 2004 9:40 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Where c
Can somebody help me with some names of good UK SIP providers?
I am looking for a UK number to connect to my asterisk
server.
Can somebody help me?
Regards,
Han
Hello,
I have a problem with connecting a Digium X100P card to a Brazilian
analog line.
Can somebody help me out with this problem?
My /etc/zaptel.conf is
loadzone=br
defaultzone=br
fxsks=1
My /etc/asterisk/indications.conf
[general]
country=br
[br]
descriptio
Is the CDR table the right table for billing?
I did some tests and CDR records billing seconds for calls
that where never picked up.
Is this a bug in my system or is that the way CDR works?
I called out on my X100T card.
Best regards,
Han
Test data
Duration 12 seconds 8 s
I have a VOIP account including a telephone number in another
country.
The connection uses the H323 connection to connect to the
remote PBX.
Can I learn Asterisk to use that connection for outgoing calls
and also that he can handle my incoming calls?
Johannes
I try to install asterisk-addons but I get an error.
Who can help me? Is my MySQL not complete or do I have another problem?
[EMAIL PROTECTED] asterisk-addons]# make install
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
cdr_addon_mysql.o cdr_addon_mysql.c
cdr_addon_mysql.c
22 matches
Mail list logo
