Can anyone tell me if I can load the modules/cdr_odbc.so module without
having to re compile my 1.4.20 production Asterisk?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix,
Any reason in particular why you don't use SIP between your Asterisk and
NexTone? This is how I have ours connected and it works well. The only
issue I've experienced is that some of the carriers that only support g729
AB have trouble with the dtmf tones from g729A, but this is not SIP
specific.
I have loaded the SIP firmware for an Avaya 4610sw IP phone and have
successfully registered it to Asterisk BE and Asterisk 1.4.18. I am however
experiencing two issues that I am hoping someone has already overcome.
The first one is that the phone looses its registration from Asterisk every
I have an Avaya 4610SW IP phone which I have upgraded to SIP firmware.
I have successfully registered this phone to Asterisk BE as well as Asterisk
1.4.18
Almost everything is working well. Except for two issues.
One of the problems is that the phone looses registration every now and
Does anyone know an easy way to disable VAD on Polycom Phones?
Thank you
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Is anyone familiar with this error message?
WARNING[26913]: channel.c:786 channel_find_locked: Avoided deadlock for
'0x82d9668', 10 retries!
Why does it happen, and how can I prevent from happening.
___
--Bandwidth and Colocation Provided by
streams of voice when you use the phone's extension as
Asterisk usually uses SIP/extension+xxx as the channel name of the call.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Ed Nuñez
Sent: Wed 9/26/2007 4:48 PM
To: [EMAIL PROTECTED]
Cc: 'Asterisk Users Mailing List
The Asterisk log file is normally located in
/var/log/asterisk
But you may want to read your asterisk.conf file to make sure the path in
which your system store it.
You will see something like this
astlogdir = /var/log/asterisk
-Original Message-
From: [EMAIL PROTECTED]
Hello list
I am having an issue with Chanspy/SIP that Im hoping someone has come
across and resolved in the past.
I am sending calls that come in TDM through T1 ZAP channels and go out to a
SIP trunk.
If I spy on the SIP channel, I can hear the person on the SIP side of the
call
Can anyone recommend a good commercial solution for queue statistics?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
What is a good solution for playing music on hold on the 1.2 branch. I do not
want to use mpg123 because last time I used it in a production server it caused
many problems. The MPG123 process was taking about 60% of my Xeon CPU.
___
--Bandwidth
if it crashes and it enables core dumps (your core
size limit is probably set to 0 when you start asterisk).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Tuesday, June 26, 2007 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED
I have installed the Asterisk BE B.2.2 image file in a new server. I need to
make network routing changes. However in their version of rPath (pound key)
Digium has removed the netconfig command. I am able to manually add the route
with
Route add default gw xxx.xxx.xxx.xxx however when I
] network routing
try to edit /etc/sysconfig/network-scripts/ifcfg-eth0 if u have eth0
if not try ifcfg-eth1 for eth1
On 6/29/07, Ed Nuñez [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
I have installed the Asterisk BE B.2.2 image file in a new server. I need
to make network
: [asterisk-users] network routing
How many GW you need to add ? if it is one .. then add
GATEWAY=xxx.xxx.xxx.xxx into /etc/sysconfig/network
thanks
Russell
On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I have installed the Asterisk BE B.2.2 image file in a new server. I need
to make
dump
Vadim Berezniker wrote:
use the safe_asterisk script
it will restart asterisk if it crashes and it enables core dumps (your
core size limit is probably set to 0 when you start asterisk).
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Ed
Nuñez
*Sent:* Tuesday, June
To configure the Cisco for RFC 2833 add the following line to the desired
dial-peer
dtmf-relay rtp-nte
Hope this helps.
Ed Nuñez
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Tuesday, June 26, 2007 11:41 AM
To: Asterisk
running, no harm done, and if it crashes, the
cron job will make sure that it's started every 60 seconds.
Any suggestions?
Thank you
Ed Nuñez
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing
I am seeing a peculiar message on my console screen on my new installation of
Asterisk 1.4.5I would appreciate any comments.
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS
Really destroying SIP dialog '[EMAIL
I have a similar issue with Qwest SIP. They only support rfc2833 in g729AB,
and Asterisk is only G729A. Sprint works fine for me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: Friday, June 22, 2007 3:21 PM
To: Asterisk
For anyone experiencing the same problem, I was able to make SpyChan work on
SIP extensions using the b and v options.
exten = _**.,1,ChanSpy(IAX2/1654|bv(4))
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nunez
Sent: Tuesday, June 19, 2007 8:05 PM
To:
some cron script with sox mixing the
IN and OUT files in 1 file .
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
Yes
This is my extensions.conf entry.
exten = _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
exten =
_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID
Just wanted to update anyone interested in this issue.
If I monitor a g729 SIP channel using ChanSpy, I am getting the same error
as when I use MixMon.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Thursday, June 07, 2007 12:14 PM
the conversation as well ?
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I installed a hardware g729 codec card in my asterisk, and I'm getting the
following error when calling from a g729 sip extension to a SIP trunk also
set to g729. The call goes through just fine, but these error messages
keep
Is anyone else having trouble going into voip-info.org today?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I installed a hardware g729 codec card in my asterisk, and I'm getting the
following error when calling from a g729 sip extension to a SIP trunk also set
to g729. The call goes through just fine, but these error messages keep flying
by until I disconnect the call.
Any ideas?
Is the autologoff function supported in Asterisk BE B.1-3? I have
configured my agents.conf with a 5 second timeout, but the agents extension
continues ringing until the call eventually goes to voicemail.
Agents.conf
[general]
persistentagents=yes
[agents]
autologoff = 5
multiplelogin = no
I had to re install the my Asterisk BE with the latest version, and when I try
to load my g.729 codec license I do not see the folders in the path that they
are described in the instructions given to us with the license or in your
online documentation. I installed the disk 1 immage (rPath),
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Monday, May 21, 2007 11:25 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FW: Re install
I had to re install the my Asterisk BE with the latest version, and when I
try to load my g.729
Does anyone know how to gain access directly to the configuration files in
AsteriskNow? I have dual NICs and need to change the binding in the config
file. I believe they blocked ssh2 access by default.
___
--Bandwidth and Colocation provided
I am having an issue with the autologoff fuction in agents.conf
I am running Asterisk BE and I am testing with agent 1656. I log in, and then
make a call into the queue. The agent's phone rings, and if I answer it, all's
fine/ I am trying to have this agent automatically be logged off
multiplelogin=no
recordagencalls=yes
monitor-join=yes
createlink=yes
updatecdr=yes
musiconhold=default
recordformat=wav49
urlprefix=http://64.211.222.226/calls/
savecallsin=/var/www/html/calls
agent = 1650,1650,Tareq Tujjar
agent = 1656,1656,Ed Nuñez
Here is my queues.conf
The g729 licenses are US$10 a pop and you can buy them directly from
www.Digium.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: Wednesday, May 02, 2007 5:10 AM
To: asterisk-users@lists.digium.com
Subject: Re Re: [asterisk-users]
Reload will reload your sip.conf file! As well as iax.conf,
extensions.conf, queues.conf, voicemail.conf, users.conf
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: Tuesday, May 01, 2007 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial
.
Thank you
Ed Nuñez
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Don
This may not be a solution to your question, but I would like to share that
Ive been having one way audio issues when connecting point to sight to a
PIX 515E using SIP. I changed to IAX and this is working perfectly now. It
was paynless to configure IAX2, so you might want to consider
Hello all
I would like to know if anyone here has had any experience trying to set up
SIP or IAX over VPN. I am testing with Cisco VPN client and when I call the
Asterisk server in my office I get one way audio.
Thanks
Ed Nunez
___
In my asterisk, I have calls coming in on a Zap channel and going out SIP.
My problem is that when I spy on the SIP channel, I hear the calling parting
breaking in and out, and the called party sounds just fine (SIP). If I spy
on the Zap channel , I hear both sides just fine. I am spying from my
to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay
I've been trying to find where to download the Web Vmail application and
instructions on how to install it for Asterisk BE. Any ideas?
Thanks
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message
Your line number nine should also specify a file name to monitor to and the
format, like this
exten = 9,2,Monitor(from-${CALLEDID}-at-${TIMESTAMP},wav)
or better yet, use MixMon instead, because this will merge the two files into
just one. (both sides of the call)
Ed Nuñez
IT/Telecom
This may be a Linux newby question, but here it goes.
I was reading the instructions on downloading and installing Asterisk GUI, but
I can't get this to work.
svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui
What would be the equivalent command in CentOS 4?
Thanks
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mail list
Sent: Friday, December 08, 2006 4:08 PM
To: Asterisk Users
Agent(1656)-caller-timestamp.wav
Thank you
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
image001.gif
Description: image001.gif
___
--Bandwidth
})
exten = s,104,Voicemail(b${ARG2})
Ed Nuñez
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Thursday, December 07, 2006 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
If you use both the public and private interfaces for VoIP in the Asterisk
Server, make sure you don't specify one of them for the binding in sip.conf
Example
bindaddr=0.0.0.0
will allow SIP traffic on any of your interfaces.
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
Singer
I would be interested to see the rest of your configuration pertaining to how
you are recording the calls. I am having trouble with this part.
Are you using monitor or MixMonitor from extensions.conf of are you using the
queues.conf or agents.conf monitor ?
Ed Nuñez
IT/Telecom
,
AGENTFILENAME=1656-20061129-183350-) in new stack
-- Executing Monitor(SIP/1656-b7d10740, wav|1656-20061129-183350-|m) in
new stack
-- Executing Queue(SIP/1656-b7d10740, NOC) in new stack
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c
-english,s,1)
exten = i,1,Playback(invalid)
exten = i,2,Goto(s,3)
exten = t,1,Goto(s,3)
#include users.conf
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
_
From: [EMAIL PROTECTED] [mailto:[EMAIL
is in such list.
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
This is how I'm able to record my outbound calls, hope this helps you.
exten = _407NXX,1,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}-OUT)
exten = _407NXX,n,Monitor(wav,${CALLFILENAME},m)
exten = _407NXX,n,Dial(ZAP/g1/1${EXTEN:0})
exten = _407NXX,n,Congestion
Ed Nuñez
IT/Telecom
Is there a way to Specify an extension number to spy on, or
monitor instead of specifying an agent or a SIP or ZAP channel?
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
It would be a SIP extension.
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Friday, November 03, 2006
3:22 PM
]
secret = XXX
permit=0.0.0.0/0.0.0.0
;deny=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Any help would be greatly appresiated
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f
= yes
port = 5038
bindaddr = 0.0.0.0
displayconnects = yes
[ami]
secret = XXX
permit=0.0.0.0/0.0.0.0
;deny=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Any help would be greatly appresiated
Ed Nuñez
IT/Telecom
I would like to know how to record all calls on a queue. Anu good sugestions?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Does anyone know how I can add a prefix to an outbound SIP call? I believe
this would be done in extensions.conf, but am not sure how to go about it.
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
From where can I download the collection of Asterisk Native Sounds?
I tried the www.astlinux.com link, but I was not able to uncompress them
because they seem to be corrupted.
___
--Bandwidth and Colocation provided by Easynews.com --
58 matches
Mail list logo
