Displaying 20 results from an estimated 200 matches similar to: "Voicemail on Different Server"
2007 Jun 01
2
asterisk mysql support
Hi all,
I've just realized that my asterisk isn't storing cdr inputs in mysql.
cdr_mysql.conf is well configured and I don't know what else should i configure.
I'm using Xorcom's packages, "cdr status" shows:
voip*CLI> cdr status
CDR logging: enabled
CDR mode: simple
CDR registered backend: csv
CDR registered backend: cdr_manager
CDR registered backend:
2007 Jun 01
2
SugarCRM Integration
Hi folks,
I was wondering if there's a guide on how to configure sugarCRM
Integration with Asterisk. I was looking in google and all i found was
about Trixbox, which has sugarcrm integrated by default.
Regards,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux Registered User #382615 - http://counter.li.org/
SIP # 1-747-633-6676 Ext. 1011
FWD # 764839 Ext. 1011
2003 Aug 04
14
Mysql CDR
hello all,
I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record.
Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault.
the original version of cdr_mysql.so works fine but I need the start time and end
2007 Jul 18
1
Error Configuring Asterisk (FREEPBX)
Hi all,
I've just installed again my Asterisk using Xorcom repositories. I can
make extensions, but when using any extension i want to dial anything,
I got "404 not found" using Xlite.
Any ideas of what can be happening?
Regards,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux Registered User #382615 - http://counter.li.org/
SIP # 1-747-633-6676 Ext.
2007 Apr 11
10
Nagios asterisk monitoring
Dear list,
I am trying to configure the nagios plugin called check_sip. I just read the
README file included with the plugin. I follow the readme steps to configure
the plugin, without success. In the nagios web interface I can see (No
output!) In the status information column. If I run the chech_sip plugin
from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u
2007 Apr 30
3
ZAPTEL PROBLEM
Hi all,
I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything
seems nice, but i'm not able to make calls nor to receive any. When I
try to make a call, I keep receiven the "all circuits are busy now"
message, and when I receive calls, asterisk doesn't seems to care
(don't get anything on the CLI)
I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's
2007 Apr 12
8
test
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<TITLE>Message</TITLE>
<META content="MSHTML 6.00.2900.3059" name=GENERATOR></HEAD>
<BODY>
<DIV> </DIV></BODY></HTML>
2006 Oct 10
2
Increase VoiceMail Messages Recording Gain - Audio Calls are Ok
Hi all
I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx,
providing Voicemail to email services for Lecagy PBX extensions.
On busy or unanswered calls, Legacy pbx will dial a specific DID (one per
extension) to asterisk, and the call is handled by Voicemail application.
I've several SIP extensions on this Asterisk box, and calls between Asterisk
extensions and legacy PBX
2007 Apr 19
1
Asterisk - Cisco Call Manager Express Trunk
Hi all,
I want to make a SIP trunk between a Cisco 2811 router and a Asterisk.
Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk
2XX). Now I want to configure a trunk so that 2811 users can call *
users. I've been reading a lot but I'm still confused.
Hope you can help me,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux Registered User
2006 Nov 23
1
Error uninstalling freepbx-panel
Hi everybody,
I've installed "future" packages (asterisk 1.2 and freepbx) from
Xorcom's Repository in a debian etch, but when i want to uninstall
freepbx-panel, i got this error:
dialer:~# apt-get remove --purge freepbx-panel
Leyendo lista de paquetes... Hecho
Creando ??rbol de dependencias... Hecho
Los siguientes paquetes se ELIMINAR?N:
freepbx-panel*
0 actualizados, 0 se
2007 May 14
0
Areski CDR
Hi folks,
I was wondering what happened to Areski CDR viewer that came before
with Freepbx. I've noticed that the live-CD contains Areski but the
repositories don't have it. Will you fix that? or shall I install
Areski from sources?
Regards,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux Registered User #382615 - http://counter.li.org/
SIP # 1-747-633-6676
2007 May 18
0
IAX2 sniffer and player
Hi all,
I was wondering if there is any IAX2 sniffer and decoder. Wireshark
can decode and play RTP streams using G.711, and Cain & Abel decodes
and plays any kind of RTP stream. But I didn't find anyone that can
decode IAX2 streams.
Any programs??
Regards,
--
Diego Quintana a.k.a. RouterMaN
Ingeniero de las Telecomunicaciones
Linux Registered User #382615 - http://counter.li.org/
SIP #
2007 Jan 12
1
eth1 with syslinux?
when will eth0 and eth1 be supported in syslinux?
--
-----------------------------------------------
Brandon Kruse
2007 May 11
1
'Invalid characters in name' with asterisk-gui
Hi all!
Is there a way to asterisk-gui to allow underline (as such cpd_tom)
in Names? It allows to [di]enable alphanumeric, but not underline noway.
Why such restriction in asterisk-gui if even asterisk users.conf allows
(and works fine) it?
Thank you,
Tom Lobato
2007 May 02
2
allowing call to my pabx every 15 minutes
Hello all,
I have a set up that answer my customer. and its working well,
however, the number of call to technical dept is what i want to reduce.
I want all call to get to voice prompt except that that enter when
minutes is 15, 30, 45, 60(in multiples of 15 minutes).
how can i achieve this and what application can i use to get this done.
I will be glad, if someone can give me a hint on this.
2007 Apr 20
2
Queue problems
Hi,
I've been configuring AsteriskNOW from the GUI but could it be that
the GUI isn't working properly? because when I make a queue and add a
few agents, and when I call the queue none of the phones ring. The
queue is also configured at "Ringall"
I checked the queues.conf file and the settings matched. I also
noticed that the agents I made in the GUI, that they were not written
2007 May 01
3
Stanaphone business ok?
I see that stanaphone is not accepting new customers. Does anyone
know if they are doing ok? I have a number with them and would like
to start redirection people before it gets canceled on me if they are
having trouble....
thanks
Todd
2007 May 04
2
question about more than one drop file
hello there all,
if i have a script that writes drop files into /var/spool/asterisk/outgoing
asterisk picks up the file and initiates the call just fine.
however, what is supposed to happen if more than one gets dropped in there
within like a second. Will it wait till the first is complete to initiate
the second ?
Do they dissapear ?
thanks
shawn
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An HTML
2010 Jun 22
1
Installing rsync-2.4.6 on an Intel box running Solaris x86....
Resending again, now that I'm a new member - AQ
________________________________
From: Quintana, Andre (M Tech Ops)
Sent: Tuesday, June 22, 2010 4:23 PM
To: rsync at samba.org
Cc: Quintana, Andre (M Tech Ops)
Subject: Installing rsync-2.4.6 on an Intel box running Solaris x86....
Importance: High
Greetings all RSYNC Guru's,
We are presently using rsync-2.4.6 on our
2007 Apr 13
2
FreePBX - Vicidial Integration
Hi all,
I am trying to install Vicidial in an existent FreePBX installation
(I'm using Xorcom packages for Debian Etch), but I didn't find any
documentation, I found only this guide [0], but is for trixbox only,
do you think it will work on FreePBX on Etch?
[0] http://iptn.org/vicidial/index.html
Regards,
Diego Quintana Cruz