Displaying 20 results from an estimated 200 matches similar to: "Virtual Question"
2008 Oct 22
0
a question about linux/asterisk/commands
hi;
I'm curious, is there a way, after i set up an admin ivr one needs a
password to enter, to make for say
option 7
for say give me current bandwith usage information? Let me tell you
what i do now and you could give me your ideas.
First of all i have a user, mbabcock that i ssh with, I'm using the
latest of freepbx and right now how i get the status for me is typing
vnstat and
2008 Nov 05
0
a callerid question
hi;
i have a customer who uses freepbx. He likes the ability to request a
caller's name and then pass that threw to the extension, call
screening, however one thing he wants to get is the ability to just
pass the callerid to the extension, verbally red out to him. Can this
be done? He's using latest build of freepbx, and i am as well. I'll do
testing on it before deploying to
2008 Dec 12
2
get first month of trixbox free
hi;
threw the end of the year we are running a promo, when ordering any
package on
http://gwhosting.net
including our vps servers and trixbox servers, you can get your first
month off. Yes, that's right, enter "30free" with out the quote signs
into the coupon code field during checkout to get your first month
free. Give us a try, you won't be sorry. Your security is our
2008 Nov 05
1
800 origination
Can anyone recommend (offlist) a good IAX or SIP based 800 provider?
Intention is for high volume calling card traffic from the US Virgin
Islands and Puerto Rico.
Thanks,
j
2008 Oct 10
2
is there a way
hey;
i'm at best western and am curious is there a way i could find out if
our best western, with out asking, is using asterisk?
oh and petsmart i think is using asterisk they have alason voice for
there main voicem enu.
mike
thanks for reading
Systems administrator and owner of http://gwhosting.net
msn: messaging at audioficks.net
twitter: http://twitter.com/creepyblindy
2008 Oct 14
3
Looking for a mentor
Looking for a mentor...
Having some issues with Asterisk 1.4.22 install. I am
new to both Linux and Asterisk, however have 20+ years
programming experience.
First off I hate asking questions I could answer
myself. I have and am reading The Asterisk manual, 2nd
edition. I have successfully installed CentOS 5.2 and
used yum to get a C compiler, current speed bump is
with ./configure
bash:
2009 Jun 11
1
[SPAM] Re: SV: HyperVM
Oh, it''s also in debian official repositories.
Ok, i''ll try it!
--------- Original Message --------
Da: "Thomas Goirand" <thomas@goirand.fr>
To: "xen-users@lists.xensource.com" <xen-users@lists.xensource.com>
Oggetto: Re: SV: [Xen-users] HyperVM
Data: 11/06/09 14:22
mattias wrote:
> yes
>
2011 Mar 24
1
Fwd: Asterisk 1.6.2.10 & CDR custom added field
Hello,
is there anyone who can point me to correct information ?
Following http://pbxinaflash.com/forum/showthread.php?t=9042 and
http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql > Extending CDR
does not result in a working environment for me.
Any feedback appreciated.
Kind regards,
Jonas.
-------- Original Message --------
Subject: [asterisk-users] Asterisk 1.6.2.10 & CDR
2009 Jun 09
46
HyperVM
Hi,
anyone here is using HyperVM?
As you probabily know, the owner of LxLabs has killed hitself (i want to
make my condolences to his family):
http://timesofindia.indiatimes.com/Bangalore/Techie-hangs-himself-in-HSR-Layout-/articleshow/4633101.cms
HyperVM has some big vulnerabilities and we don''t know if they will be
fixed. We don''t know if the licensing server will be kept
2009 Jul 28
1
outbound calls not reaching vitelity
Any vitelity customers with pbxinaflash boxes? I'm able to call
in-house, but failing to make outbound calls. My assigned server at
vitelity is not reachable. I can ping to my ISP OK.
Any help appreciated. Such as actually how to make email contact with
support at vitelity. They're not responding.
Thanks, Tom
2010 May 17
1
Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk?
Hi Guys,
I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the
current 1.4.10 version. I am running Asterisk 1.4.x (in fact it is a
PBXinaFLASH system).
How can I upgrade to the latest Libpri? Do I need to re-install Asterisk?
Won't that break the box?
Can I simply do this to upgrade:
*rm /usr/src/libpri/*.**
*rm -rf /usr/src/libpri/**
Download the new libpri and put
2010 Oct 29
1
Updating asteriskcdrdb with uniqueid field from Master.csv, Master.csv.1....Master.csv.5 - Must I bring all files together first?
Hi Everyone,
Just noted that PBXinaFLASH failed me again on yet something else. The
uniqueid field didn't update on the asteriskcdrdb in the past few months but
it is available in the .csv files in /var/log/asterisk/cdr-csv/*.csv
I have already re-did the asterisk-addons install with correct parameters to
include future calls "uniqueid" in the table (I have no clue why these
2008 Nov 17
2
is udev necessary?
Hi all
I recently setup a CentOS 5.2 server, running XEN (using HyperVM), and
then moved the hard drive from my test box to my Intel server.The
problem I now have, is that it doesn't bootup properly. Shortly after
I see the udev service started, the machine reboots. This keeps on
going the whole time.
I have managed to kill udev on start-up (with CTRL + C), and then it boots up.
So, do I
2010 Apr 19
1
Zap PRI failed with Cause 34 - Where to check for problems?
Hello Everyone,
I have a system that was working on Sunday 1 P.M. and then gives Congestion
on Monday morning. Sometimes over night it probably stopped working. It's a
PBXinaFLASH with Asterisk 1.4 and libPRI with a 23 channel PRI connected and
24th D-Channel.
This is all I see in /var/log/asterisk/full:
[2010-04-19 08:45:50] WARNING[29707] app_dial.c: Unable to create channel of
type
2009 Jan 29
0
Benqtelecom in cdr log
hi all,
I've been noticing some unknown traffic in my cdr logs approximately 50
quantity like this;
2009-01-29 08:17:24 SIP/96.9.1... BenQTelecom "BenQ Telecom" s ANSWERED
00:20
I wonder that have anyone ever faced like this problem ?
I think it's a security problem, when I googled I have found one topic
explanation in english
2010 Mar 23
0
Classic NO AUDIO problem - DD-WRT and NAT forwarding - HELP PLEASE!
Hi Everyone,
I have tried to set the box to DMZ and also tried to port forward 5060
TCP/UDP and 10000/20000 UDP to the server IP but it's no use and there is a
no audio issue. I am pretty certain it's a NAT issue as the sip call
establishes. I also made a succesful IAX2 call through IAX trunking and zap
lines on this server but sip doesn't work. I register to an extension but
even
2010 May 28
1
"pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11
Hi Guys,
I am running a PBXinaFLASH server. I replaced contents of /usr/src/libpri
with the new version of Libpri v1.4.11. The installed one was v1.4.10.
System is running Asterisk 1.4.21.2.
I did the following after:
cd /usr/src/libpri/
make
make clean
make install
Install end with these lines.....:
*ln -sf libpri.so.1.4 libpri.so*
*mkdir -p /usr/lib*
*mkdir -p /usr/include*
*install -m 644
2010 Jun 20
0
Deleting some of the CDR data - How to do it safely?
Hi Guys,
I am looking to delete some of the CDR logged by Asterisk in asteriskcdrdb
in a PbxinaFlash system running Asterisk 1.4.x
The CDR records to deleted are probably a big chunk and spread out all
through the database but I basically want to delete all calls that came in
through a specific DID. I think they all show as SIP/did_number
I don't want to break the system or break the
2011 Mar 22
0
Asterisk 1.6.2.10 & CDR custom added field
Hello list,
I have added an extra field "mycolumn" to the cdr table in my MySQL-DB.
I simply try to add a value to this column by doing the following in the
dialplan :
exten => 600,n,Set(CDR(mycolumn)="myvalue")
But this value is not written to the column 'mycolumn' together with the
other CDR-data.
Why is this ?! Do I need further configuration ? (Not
2009 Mar 16
2
Multi-tenant with receptionist features for managed service
Dear all,
I'm currently researching options for a MT asterisk gui/system for a
small business centre that will have 12 units in it. Each unit will be
configured for one extension.
The system there will have a max of 12 concurrent calls to PSTN
provided via an ADSL/SDSL link to our VoIP provider in the UK, using
g.711, maybe g.729 dependant on networking costs. Fallback will
be to 4 analogue