Displaying 20 results from an estimated 2000 matches similar to: "call Intrude"
2004 Sep 03
1
zap barge restrictions
I have a couple of questions on the zapbarge:
1) zapbarge asks for a channel - how would a manager know what channel to
enter ? Is there any way of being able to enter an extension number instead
? I know that you can get the information from the manager interface, but I
wouldn't want to give my users access to this, or have to install / write a
system just to get an extension number from a
2004 Jun 11
3
Background Playback fails
Hi Guys.
I've had a lay off from Asterisk for 12 months but I am starting to look
into it again. I am not very Linux savvy and found it hard going the
last time. I've started playing with it in the last 3 weeks and I have
to admit to making more head way this time.
The first problem I'm stuck on and I cant find a solution to is that
sound files that I have recorded (be it by
2008 Nov 18
1
How to Barge specific extensions
Hi All
Can anybody help me for dial plan to barge or Spy(ExtenSpy)
specificor selective extemsions among 20 extension in my office.
lets say my office extension range is 301-320 & i want to barge only 3
extension say 320, 302,314.
is this possible to barge specific extension? . Plz help me for this.I
am using Asterisk 1.4.9 & SIP channels.
Regards
Amit
-------------- next
2007 Oct 02
2
Having problems posting to the list
Hi All
I'm having problems posting to this list, no bounces the mails just
dont show
any advice how to get the postings through is there filtering?
robb
2003 Nov 05
6
recording calls
Hello,
You can use ZapBarge as an extension in your dialplan to listen in on
conversations going on in Zap channels(Zaptel device channels)
As for recording you can use the Manager interface command StartMonitor to
start recording of a Zap channel and StopMonitor to stop it.
Zap channels are pretty much the only ones right now that you can directly
monitor and record through Asterisk.
If
2008 Nov 30
3
DTMF Tones
Hi All
I cannot seem to find a way to stop atserisk inercepting DTMF tones and
regenerating them even on a zap to zap bridged call
is this possible?
Thanks
Robb
2004 May 13
3
recommend a Linux based TFTP server
Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box?
Thanks in advance
Robb
2003 Nov 06
5
FW: recording calls
Sorry that got accidentally sent incompleted, here's the full post:
OK, here is the long drawn out description of how I am using Zap Barge and
Monitor:
Zapbarge(listen in on live calls):
Very simple actually I just added this to my dial plan(extensions.conf):
; barge monitoring extension
exten => 8159,1,ZapBarge
exten => 8159,2,Hangup
Then when you dial 8159 on
2007 May 11
2
megasr Sata Raid driver and the lastest kernel
Hi List
I'm trying to update to the lastest kernel but I have a dirver that is not
inculded in the distrubution, and I had to use the driver disk when installing
centos 4.4 in the first place, The driver megasr .ko works fine with the
installed kernel but I cannot find on for the updated kernel, any adive would
be appreciated.
without the updated driver there is a kernel panic on boot due to
2007 Apr 17
1
Transfercapability DIGITAL
Hi
I have a requirement to bridge Digital ISDN call through an asterisk box
but no matter what I setup in the dial plan the second leg of the zap
bridge is always set to Transfer Capability of SPEECH, I wondered if any
one has come across this and managed to fix it?
Thanks in advance for your help
Robb
2004 Jun 12
9
Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
How do you prepend. I want to be able to dial 7 digits instead of of
11 for local calls.
Can someone post there extensions.conf part that is relavent?
2004 Jun 12
5
MWI on Cisco ATA-186 (SIP)
I am trying to set up the Message Waiting Indicator (stutter
tone/light) so that my cisco ata-186 will let my phones know there is a
message waiting. However this does not seem to be very well
documented.
I found this on wiki mailboxnumber@context ... where does that go? Do
I put it in my SIP.conf definition for my cisco ata, or where. In my
SIP cisco definition i already have a
2004 May 27
4
AGI Pascal
Hi,
Has anyone done any AGI scripting in pascal. I would appreciate help anyone
can offer. My understandin on AGI scripting is very flaky, I am assuming
whatever language is used the application needs to be compile and made
executable. So if I write a script in pascal, I would compile it with
something like freepascal and make it executable.
Thanks
Umar Sear
2004 Jul 28
2
Asterisk voicemail from mysql no longer working
Hi All,
I hope someone can help.
I have a system that I have recently upgraded to
latest CVS and my voicemail is not working from mysql
database.
I get an error on the console saying
" No entry in voicemail config file for 'number'"
whilst there is an entry in the database for the
specified number. It seems like app_voicemail is no
longer checking the database even though
2008 Nov 20
2
ISDN Cause codes
Hi All
Just been looking at stats for one of my sites, and I'm conserned about
the number of error cause codes being returned from the telco
for example
12000 calls processed
131 are cause code 31* normal. unspecified.*
139 are cause code 28 * invalid number format (address incomplete).*
112 are cause code 1 *Unallocated (unassigned) number.
*this adds up to about 3% of calls not
2004 Jun 11
3
ssh key problem
Hi I've need to reinstall my asterisk software (hard drive failure). I'm
back and running to a make samples state.
I have backed up all of my conf files (ok so they were about a week old
but much better than starting from scratch), the problem I am having is
with WS_FTP Pro.
Basically I used to connect to my asterisk server using this software no
problems just using root as
2003 Apr 19
7
Call screening
I've set up asterisk with my X100P as a home answering machine. Works great
so far - answers the phone after 20 seconds, runs the phone tree, emails
voicemail, etc.
However, the one feature traditional answering machines have that I haven't
been able to figure out is how to listen in on the call. Ideally I could
just route through Console/dsp and hear it on my speakers. I've tried
2004 Oct 05
5
Asterisk Perl AGI
Hello everybody:
This could be a stupid question, or may be not; I'm not sure 'cause I have not a very wide experience working with Asterisk, actually I just started last week. I need to make an IVR system work and I choose working with AGIs, written in Perl.
The available documentation I've found show it as a very simple proccess, but it doesn't work for me... and I
2003 Sep 12
1
Dect Phone
Hi
I have a problem with a new DECT phone I have bought
The key pad works like a Mobile phone where you dial first then pick up
the line, but it seems to dail too fast or spuriously, ie 012826736464
show on thew Asterisk console as 0012282677, could any one offer advice
how to fix?
Also when doing a ZAP bridge to this phone from an outside line the call
is very echoy, but not an internal