Actually I registered two users in my X-lite. Both the users registered in
different asterisk servers. While calling, first you have to right click on the
x-lite and the click on the required server. Then make call. It will work.
-Original Message-
From: [EMAIL PROTECTED] on behalf of
Hans Witvliet [EMAIL PROTECTED] writes:
On Sun, 2008-01-27 at 20:19 +0100, Benny Amorsen wrote:
You can't have a USB handset without a soft phone. You can get some
which automatically run the phone software when plugged in, but that
only works in Windows.
How about a udev rule?
Sorry, I
Tzafrir Cohen [EMAIL PROTECTED] writes:
Asterisk (chan_alsa, chan_oss, chan_console) is a soft phone.
True, but deploying Asterisk to user PC's is not a particularly
attractive option.
/Benny
___
-- Bandwidth and Colocation Provided by
Hi,
I am trying to utilize MWI with sip channel.
when my client sens a SUBSCRIBE to asterisk I get info that user not found:
-
[Jan 28 11:49:02] --- (19 headers 0 lines) ---
[Jan 28 11:49:02] Creating new subscription
[Jan 28 11:49:02] Sending to 192.168.129.38 : 7060 (no NAT)
[Jan
On 1/25/08, Raj Jain [EMAIL PROTECTED] wrote:
Hi,
I'm trying to implement a Voice Drop service within Asterisk
dial-plan. The service is supposed to work as following:
1. A initiates a call to B
2. The call is answered by B's answering machine
3. A hears the answering machine's greeting
Hi all,
I'm using asterisk to provide a simple service for official time (hour +
minutes + seconds).
The system and application (asterisk + zap detection + custom application) is
monitored by Nagios with some scripts I have created using examples from
voip-info.org.
But I still need to
On Monday 28 January 2008 14:00:27 Rahul Yadav wrote:
Hi all
i am getting a serious problem.I am using asterisk 1.4.15 and dialing
outbound through sip.
The problem is that whenever i dial a number the other person can hear my
voice but i dont hear anything.
Have you tried:
Hi all
i am getting a serious problem.I am using asterisk 1.4.15 and dialing
outbound through sip.
The problem is that whenever i dial a number the other person can hear my
voice but i dont hear anything.
help me
Thanks
Rahul
___
-- Bandwidth and
I use mrtg,
I call this from MRTG: `/usr/local/groundwork/nagios/libexec/asterisk-mrtg.pl
-1 Zap -2 SIP`
Here is my asterisk-mrtg.pl: (note, I have tweaked this, but I do not know
where I had gotten my original reference)
---
#!/usr/bin/perl -w
use strict;
use IO::Socket;
use
Hi,
when I'm trying to call the following extension
exten = 6002,1,Verbose(1|Extension 6002)
exten = 6002,n,Dial(Agent/6002)
exten = 6002,n,Hangup()
the call is terminated and I get the following warning from asterisk:
app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent'
zttest:
--- Results after 44 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.988350
Dell 2950
Wildcard TE410P (2nd Gen) (rev 02) (with echo cancel daughter board)
1 PRI configured to Telco
1 PRI configured to old Panasonic DBS 576 being used just as a mux for our fax
machines.
On 1/28/08, Thomas Kenner [EMAIL PROTECTED] wrote:
Hi,
when I'm trying to call the following extension
exten = 6002,1,Verbose(1|Extension 6002)
exten = 6002,n,Dial(Agent/6002)
exten = 6002,n,Hangup()
the call is terminated and I get the following warning from asterisk:
app_dial.c:1106
JR,
That script runs fine. You should be able to run it first manually, if so
please copy and paste the error.
Thanks,
Alejandro,
After reading the sparse info and attempting to get this running, I'm
unsuccessful and could use some guidance.
I already have a MRTG server up and running
After reading the sparse info and attempting to get this running, I'm
unsuccessful and could use some guidance.
I already have a MRTG server up and running serving hundreds of router
interface graphs. I would like to add SIP/IAX channel graphs for all
our asterisk servers. I'm running
On Mon, Jan 28, 2008 at 04:37:25PM -0200, Pablo Allietti wrote:
hi all i have a te110p installed in my system with a lot of Echo..
i decide to install the oslec echo supressor but when y try to add the
module i have this problem.
[EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko
hi all i have a te110p installed in my system with a lot of Echo..
i decide to install the oslec echo supressor but when y try to add the
module i have this problem.
[EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko
insmod: error inserting 'wct1xxp.ko': -1 Unknown symbol in module
[EMAIL
On Thu, 24 Jan 2008, Steven wrote:
I use mrtg,
I call this from MRTG: `/usr/local/groundwork/nagios/libexec/asterisk-mrtg.pl
-1 Zap -2 SIP`
Here is my asterisk-mrtg.pl: (note, I have tweaked this, but I do not
know where I had gotten my original reference)
Hi Steven,
The original
On Monday 28 January 2008 14:10, Steve Totaro wrote:
On Jan 28, 2008 1:33 PM, Ron Joffe [EMAIL PROTECTED] wrote:
I have a system with 2 TE220B (2xPRI). I am looking for a method to
shutdown one of the 4 zap lines.
Something like a ifconfig eth3 down command.
It should be the
I have a system with 2 TE220B (2xPRI). I am looking for a method to shutdown
one of the 4 zap lines.
Something like a ifconfig eth3 down command.
It should be the equivalent of physically unplugging one PRI circuit.
___
-- Bandwidth and Colocation
JR Richardson wrote:
6. Re: That script runs fine. You should be able to run it first
manually, if so
please copy and paste the error.
mrtg:/var/www/mrtg# env LANG=C /usr/bin/mrtg
2008-01-28 11:16:01: WARNING: Could not get any data from external
command '/var/mrtg/10.10.14.102.cfg
6. Re: That script runs fine. You should be able to run it first
manually, if so
please copy and paste the error.
mrtg:/var/www/mrtg# env LANG=C /usr/bin/mrtg
2008-01-28 11:16:01: WARNING: Could not get any data from external
command '/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2
this is the output of dmesg
[EMAIL PROTECTED] zaptel-1.4.7.1]# dmesg |tail
wct1xxp: disagrees about version of symbol zt_transmit
wct1xxp: Unknown symbol zt_transmit
wct1xxp: disagrees about version of symbol zt_rbsbits
wct1xxp: Unknown symbol zt_rbsbits
wct1xxp: disagrees about version of
this is the output of dmesg
[EMAIL PROTECTED] zaptel-1.4.7.1]# dmesg |tail
wct1xxp: disagrees about version of symbol zt_transmit
wct1xxp: Unknown symbol zt_transmit
wct1xxp: disagrees about version of symbol zt_rbsbits
wct1xxp: Unknown symbol zt_rbsbits
wct1xxp: disagrees about version of
Unknown option: h
Unknown option: 1
Unknown option: 2
ERROR: Line 3 (use strict;) in CFG file (10.10.14.102.cfg) does not make sense
I named the script file the IP address of the server.cfg instead of
asterisk-mrtg.
I call the script from the command line:
# env LANG=C /usr/bin/mrtg
On Jan 28, 2008 1:33 PM, Ron Joffe [EMAIL PROTECTED] wrote:
I have a system with 2 TE220B (2xPRI). I am looking for a method to shutdown
one of the 4 zap lines.
Something like a ifconfig eth3 down command.
It should be the equivalent of physically unplugging one PRI circuit.
With or
I have a system with 2 TE220B (2xPRI). I am looking for a method to shutdown
one of the 4 zap lines.
Something like a ifconfig eth3 down command.
It should be the equivalent of physically unplugging one PRI circuit.
___
-- Bandwidth and Colocation
Hello list, hope some one could help me find the answer.
Asterisk 1.4.16.2 installd as no-root user
The main issue is that every now and then, cd * box seems to loose the
user's registrations, there is nothing in the console, absolutely no
messages, only when another friend trys to dial an
Hello there -
I have an AVM C4 card, which has 4 controllers. I am able to dial
from outside to SIP phones - but not to ISDN phones. (an from SIP to
the outside world)
I want to have Asterisk betwen the PSTN and the internal ISDN phones.
So, controller 1 of the c4 card is connected to
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug
On Mon, Jan 28, 2008 at 05:07:54PM -0200, Pablo Allietti wrote:
this is the output of dmesg
[EMAIL PROTECTED] zaptel-1.4.7.1]# dmesg |tail
wct1xxp: disagrees about version of symbol zt_transmit
wct1xxp: Unknown symbol zt_transmit
wct1xxp: disagrees about version of symbol zt_rbsbits
On Mon, 2008-01-28 at 18:05 -0500, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this
On Thu, 2007-06-28 at 22:37 -0500, Russell Bryant wrote:
Bent Bagger wrote:
When will these additions make their way into the Asterisk mainstream
It has not yet been merged into the main development tree, but I'm sure it
will
be before Asterisk 1.6 is released.
Any progress on IPv6 ?
Hello List,
I am currently having a bit of a strange issue with a pair of asterisk servers
that we recently set up.
For a bit of background, this particular business has two sites in two
different towns, about 10 minutes apart. They have 3 analogue PSTN lines
connected to the asterisk servers
Too much info then too little info.
Basically the issue is the provider this happens even when we send
them the calls in IAX because they talk SIP to the same gateway.
I just need to prove it to these people. Anyone have any DTMF issues
between Asterisk and a Quintum gateway?
On Jan 28, 2008
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Followed the instructions at
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
I dead end at patching the channels Makefile. There have been some
changes since these instructions were made. I added the chan_unicall.c
to the channels folder but
I think your best bet is to do a packet capture and look for RTP packets
with an RTP Event payload (rtpevent display filter).
On Mon, 28 Jan 2008, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global
On Mon, 2008-01-28 at 17:03 -0700, James Finstrom wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Followed the instructions at
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
I dead end at patching the channels Makefile. There have been some
changes since these instructions were
Does turning off the notransfer help? I would imagine that dropping the
second server out of the equation might be useful, and save some
bandwidth.
PaulH
On Tue, 2008-01-29 at 10:38 +1100, Daniel Cole wrote:
Hello List,
I am currently having a bit of a strange issue with a pair of
Hans Witvliet wrote:
Any progress on IPv6 ?
Still completely seperate code, or is it already being merged into the
tree...
Perhaps i overlooked it, but i couldn't find any reference in:
http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co
There has been progress. It is not
The Asterisk development team has released versions 1.6.0-beta2 and and
1.4.18-rc1.
The new beta for 1.6 is available for download from
http://downloads.digium.com/. The release candidate for 1.4.18 is only
available via svn. It is available for anyone that would like to help test
1.4.18
Try cyber-telecom.net
May be get a X100P with a CT-G1000 or G2000
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Tuesday, January 15, 2008 3:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
What does 'show agents' give you? 'show queues' would be useful too.
PaulH
On Mon, 2008-01-28 at 13:36 +0100, Thomas Kenner wrote:
Hi,
when I'm trying to call the following extension
exten = 6002,1,Verbose(1|Extension 6002)
exten = 6002,n,Dial(Agent/6002)
exten = 6002,n,Hangup()
Try cyber-telecom.net
May be get a X100P with a CT-G1000 or G2000
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Sunday, January 20, 2008 11:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
It is more economical to get a hardware GSM Gateway from places like
cyber-telecom.net and then plug it in a X100P
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: Monday, January 14, 2008 8:43 PM
To:
On Jan 20, 2008 4:40 PM, Michael Graves [EMAIL PROTECTED] wrote:
I'd like to add a device to my Asterisk server to leverage my cellular
account. Does anyone on-list have experience with hardware gateways vs
using cah_bluetooth and an old cell phone?
We use the Junghanns.NET duoGSM PCI card
On Tue, Jan 29, 2008 at 09:12:11AM +0800, Sam Tam wrote:
Try cyber-telecom.net
May be get a X100P with a CT-G1000 or G2000
a) this should be on the biz list
b) why don't you post from your cyber-telecom.net address?
c) it must be the end of the sales cycle and trying to get a bit more
revenue
With that sort of set up, If for example i get a 8 channel GSM gateway
and the X100P can i make more than 1 concurrent call though the gateway
with the X100P or does it only support 1 call at a time?
What im looking to do is get a multi channel GSM gateway, and have the
ability to make more
Not to be out of topic but gmail is really good at managing mailing list
that why I use gmail.
Also it can filter out spam and my cyber-telecom website is not very good at
doing that..
I am trying to offer a solution to a person who want a solution. And if that
is a bit too much then next time
You need to take a step back and first test the script without using
MRTG. Execute it like this:
# /opt/bin/asterisk-mrtg -h localhost -u XXX -p -1 SIP -2 Zap
10
10
10
10
You should get 4 lines of numbers. That respresents your SIP and Zap
channels. Once you get past this step
Hello Kev
There are 2 solutions for this actually there are more but I don't want to
make things too complicate this time.
1. get a 8 ports fxo card for asterisk. There are ebay and many other on the
asterisk-biz list that sell them or me of course then get a 8 ports gsm
gateway and plug those
On 1/28/08, JR Richardson [EMAIL PROTECTED] wrote:
You need to take a step back and first test the script without using
MRTG. Execute it like this:
# /opt/bin/asterisk-mrtg -h localhost -u XXX -p -1 SIP -2 Zap
10
10
10
10
You should get 4 lines of numbers. That
Why not give the receptionist a two line phone? Register one line on
server 1 and the other on server 2. Then the bounce back and forth goes
away saving bandwidth.
Lyle
Daniel Cole wrote:
Hello List,
I am currently having a bit of a strange issue with a pair of asterisk
servers that we
Hi list,
Anyone knows where I can get information about configuring a Dialogic
card to run with Asterisk?? The model I have is D/120JCT-LS. Somebody
told me that I had to buy the driver, but I don't know if this is true
and if so, who, how and how much...
On Tue, Jan 29, 2008 at 10:44:36AM +0800, Sam Tam wrote:
Not to be out of topic but gmail is really good at managing mailing list
that why I use gmail.
Also it can filter out spam and my cyber-telecom website is not very good at
doing that..
I am trying to offer a solution to a person who
Hi All,
PLEASE READ if you depend on Asterisk CDR's and support transfers.
Apologies for the shout but I'm desperate to get others to agree Asterisk has a
big problem with the CDR's that are generated for transfers. I can understand
why not too many people are interested as transfers are
Hi,
Do you know http://www.moythreads.com/astunicall/ ???
Luis A P Barbosa
2008/1/28, James Finstrom [EMAIL PROTECTED]:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Followed the instructions at
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
I dead end at patching the channels
So here is my setup.
Hardware:
Intel P3 1.2 Ghz
1 GB RAM
36 GB Drives Mirrored
Software:
CentOS 5
2.6.18 Kernel
Asterisk 1.4.14
Zaptel 1.4.7 (redfone)
LIbpri 1.4.2
I'm using TDMoE with my PRI using a product called fonebridge from a company
called redfone. They require that I use their own
Thanks Paul and Lyle for the suggestions.
I would like to keep the phones configuration to one line for now, and see if I
can solve the problem rather then just work around it.
I have changed he notransfer option, will see what happens over the next few
days.
Thanks again for the suggestions,
What if you issue restart now and then ctrl-c or ctrl-d out of Asterisk?
IMO TMDoE support is very legacy I don't think its really been
maintained since the 1.0 builds.
On Jan 29, 2008 1:24 AM, Mark Greene [EMAIL PROTECTED] wrote:
So here is my setup.
Hardware:
Intel P3 1.2 Ghz
1 GB RAM
36
Hi,
I am trying to install Gatekeeper. I have installed pwlib and trying to install
h323 plus.
I have set the path as
PWLIBDIR=/root/pwlib
export PWLIBDIR
OPENH323DIR=/root/open323
export OPENH323DIR
LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib
export LD_LIBRARY_PATH
I also configured
broadband Voice ha scritto:
Irqbalance was causing the the processor handling the interrupts of
the zap cards to change very often.
This would impose a delay during the change and cause the zttest
numbers to drop/be inconsistent.
Irqbalance is a good idea BUT some kernel
Steven ha scritto:
zttest:
--- Results after 44 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.988350
Dell 2950
Wildcard TE410P (2nd Gen) (rev 02) (with echo cancel daughter board)
1 PRI configured to Telco
1 PRI configured to old Panasonic DBS 576 being used just as a
broadband Voice ha scritto:
Does anyone have any compatibilty issues with Dell *PowerEdge^TM 2950 III
2-Socket, Quad-Core 2U*? I plan on using this with the Digium T1 cards.
Thanks.
Consider 2 socket dual core CPU with more mhz, Asterisk is more IO
than computation. The quad core CPU is
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