Nigel,
I read your post, because i have a new set up Centos 4.4, kernel
2.6.9-42.0.2.EL, yum updated and no digium hardware.
But, I cannot load the the modules zaptel and ztdummy.
I follow all the directives from voip-info.org but i cannot get it works.
May you help me?
Kind regards,
Juanjo
Dear Friends,
One customer of mine has a line from Vonage connected to his Asterisk
box, he receive the following messages each 30 seconds in the CLI:
REGISTER attempt 1 to [EMAIL PROTECTED]
May you help me to understand what it means and how can I avoid this messages?
Thank you in advance,
Dear Colleagues,
I have 2 phones Cisco 12SP+ connected to my asterisk box 1.2.6 and
SCCP channel version: 20060408.
When a call is generated no audio pass through the phones, neither if
i call from a 12SP+ to another nor calling between 12SP and other
phone (ex. an x-lite). Only sometimes works
Hi,
After reading this valuable forum and the voip-info wiki and follow
all the steps , but my Cisco 12SP+ remains unregistered.
These are my config files:
skinny.conf
[general]
port = 2000 ; Port to bind to, default tcp/2000
bindaddr = 172.20.1.1 ; Address to bind to
Derek,
Thank you for your quick response.
If i do a skinny debug, nothing happens :(
And this shows my CLI:
server*CLI skinny show lines
101 1 101N N
server*CLI skinny show devices
Name DeviceId IP
Thank you Traci,I put this two variables in my .conf file and it works!!!Well, It seems that this variables are not necessaries in old versions, but in newest ones.Thank you again,Juanjo
I believe you are missing 2 variables in your conf file:table=cdr(the table your cdrs should be
Dear Tomislav Colleagues,
I read your post in the Astersik-list about you can not compile the cdr_mysql on Asterisk 1.2
Well, I have a similar problem.
I compiled the cdr_mysql module, but when i start Asterisk the following error appears:
Dec 12 18:03:33 WARNING[7237]: loader.c:325
My cdr_mysql.conf is the same I was using for version.1.0.9 and it is as follow
[global]
hostname=localhost
dbname=dbasterisk
password=dbpassword
user=dbuser
userfield=1
Any ideas?
Thank you in advance,
Juanjo
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Dear Collegues
I am trying to compile the new version (Asterisk.1.2) with my debian
box and i get the following error when i compile the zaptel package:
radio:/usr/src/asterisk-1.2/zaptel-1.2.0# make
make: Warning: File `Makefile' has modification time 3.1e+08 s in the future
cc -I. -O4 -g -Wall
Dear colleagues,
I read on the web that implementations of thousand SIP extension on
asterisk became worse and people suggest SER + asterisk.
Anybody checks this? what is the problem?
Any comment
Kind regards,
Juanjo
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Dear Colleagues,
I Have my * with a X100P clon card. When a call in from the PSTN and
nobody answer the call go to the voicemail, then the caller my hangup
or press #. If the caller hangup the ZAP channel never hangup, but if
the caller press # the ZAP channel hangup. Even every time the outside
Dear Sirs,
I have an Asterisk box with two X100P one connected to the PSTN and
the other one with a conventional PBX.
When the asterisk pass a call from the PSTN to the PBX (making a zap
channel bridge) the Zap channel is never realised. Neither when the
PSTN hangups nor when the PBX hangups.
a * before dialing. You can setup
your dialing prefixes to add it automatically so it becomes transparent to
users.
Pat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juanjo Portela
Sent: Friday, 13 May, 2005 19:07
To: Lista Asterisk
Subject
Sirs,
Thank you for your quick response.
But when i try to make a call to FWD the following error appears:
For example, when i call to 612 (a service number of FWD)
-- Executing Dial(SIP/Phone4-e85b,
SIP/[EMAIL PROTECTED]|90|Ttr) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP
Dear Sirs,
I was using iaxtel to make calls to 1-800 phones for free, but
unfortunatelly it is no working ...
Anybody knows who may give me similar service than iaxtel?
Thank you in advance,
Juanjo
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Great Steven !!!
Thank you very much. I installed the libtiff-devel package and all run
peacefully !!!
KInd regards,
Juanjo
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Asterisk-Users@lists.digium.com
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To
Sirs,
I can't compile the source spandsp-0.0.2pre10; when i try to do the
make sentence the following errors appear:
# make
Making all in src
make[1]: Entering directory `/export/usr/src/spandsp-0.0.2/src'
make all-am
make[2]: Entering directory `/export/usr/src/spandsp-0.0.2/src'
if /bin/sh
] Problem Compiling Spandsp
From: Juanjo Portela [EMAIL PROTECTED]
Date: Mon, March 14, 2005 6:44 pm
To: Lista Asterisk asterisk-users@lists.digium.com
Sirs,
I can't compile the source spandsp-0.0.2pre10; when i try to do the
make sentence the following errors appear:
# make
Making
Dear Sirs,
I had compiled PWlib and OpenH323 correctly in my Fedora Core 2.
But when I try to compile asterisk-oh323 I get the following error:
`__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__'
How can I solve it?
Thank you for your help.
Juanjo
Mario,
I solved this problem making teh following symlink :
ln -s /lib/modules/2.6.8-1.521/build/ /usr/src/linux-2.6
I hope this may help you.
Regards,
Juanjo
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