Displaying 20 results from an estimated 3000 matches similar to: "BLF in Asterisk 1.4.*"
2010 Oct 30
8
Under heavy attack
My main asterisk server is under unusual heavy attack, and so far Fail2Ban
has blocked about 30 IPs, from various different countries. At this time it
is blocking about 1 IP address every few minutes.
Just wondering if anybody else is also experiencing unusually increased hack
attempts today?
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
-------------- next part
2010 Nov 01
4
FW: Under heavy attack
Only 100? We had a single server over 300.
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria
Sent: Saturday, October 30, 2010 9:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Under heavy attack
My count has reached 100 for the day. The server serves doesn't serve
2010 Oct 23
3
Cepstral voice quality not good
Hello list,
I have been using Cepstral's 8KHz voices for my text-to-speech service for
some time now, and have been noticing that the voice quality is really poor,
doesn't matter what phrase I give it to convert. None of the other 8KHz
voices I have ever used were this bad. It doesn't seem good enough system to
be used in a commercial system. Is there any better quality text-to-voice
2010 Dec 15
2
Recommendation for a Linux based SCADA
Hi list,
For a telecom project I need to setup a SCADA solution. I don't have any
previous experience in this type of monitoring and automization. I'll be
using SNMP data from asterisk servers and endpoints. If anybody has any
suggestion which SCADA software can fit in such a VoIP solution, your
guidance will be highly appreciated.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
2011 Jan 21
4
Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
Hi list,
For a client I am setting up a system which will use T1 PRI from Primus, who
offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only
used switchtypes euroISDN and National. Although the documentation says
Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you
have used these protocols on an Asterisk box and if there were any things to
consider. If
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2010 Aug 25
6
AEL - what is error: ael.flex:647 ael_yylex: Unhandled char(s):
Hi List,
When doing 'ael reload' on two servers, which are setup with asterisk 1.4.22
and 1.4.35 respectively, I am getting multiple lines of this strange error:
ERROR[15483]: ael.flex:647 ael_yylex: Unhandled char(s):
On three other servers with same versions of asterisk, i.e. 1.4.22, I don't
see this error.
Number of lines of the error are the same as the number of lines of the
2010 Jul 14
2
BLF with Realtime
Hello Asterisk community,
I'm trying to use BLF with Asterisk Realtime, i've been searching for
some info but nothing seems to be clear, can anyone help me eith some
ideas to make this work ok?
I'va my dialplan with Realtime
Thanks in advance
--
Saludos
Danny Dias
SkypeID: danny.dias1
2009 Dec 05
2
How to use SIP hints and BLF for realtime extensions on Aastra phones?
Hi,
I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk
1.4 using realtime architecture. Extensions are defined in realtime database
and dial plan is in AEL. I am able to correctly setup hints in the dialplan,
but they don't work. Did some research and found out that hints don't work
work with realtime extensions. Is there any work around?
On voip-info I read
2010 Oct 26
2
No media being sent in SIP call
Hi all,
I seem to be having a strange problem with a sip trunk.
On a fairly frequent basis, I'll make a call, ore receive a call, and there
will be NO sound. The strange part is that both endpoints are public IP
addresses so NAT isn't in play and a sniffer trace reveals that the packets
simply aren't being sent.
It only seems to happen on a particular trunk. The same phone
2010 Nov 05
2
Determine channels in use from CLI
Is the a CLI command that shows all channels in use at one time? (Whether IAX, SIP, SCCP, etc)?
As well, when I "SIP SHOW CHANNELS" I see phones registering showing as channels in use. Is there a way to filter this output?
Thanks!
MD
2010 Oct 23
7
Dial plan help
Hi,
I am facing issue while generating a dial plan for the following case:
all caller should be asked a code to enter than All the callers should be
connected one extension.
also tell me testing scenario :
I have pbx setup and currently I have soft phones to use as extension.
Currently I have created a dial plan using vdp I tried submitting it here
but I don't know how to extract text
2010 Aug 24
9
Should I move to 1.6 or 1.8, or stay with 1.4?
Hi list,
I am planning a migration to virtual machines, and was considering with it
to move from 1.4 to one of the later versions. My and my clients' 1.4 setups
have been rock solid and I don't want to put myself into any unnecessary
trouble. Those of you with solid experience with all these versions, what
would you suggest? What new and exciting enhancements would newer versions
bring
2010 Nov 02
2
Ring Freq
Hi
I'm sorry for the my trivial quest.
I Have asterisk 1.4 with TDM 400 with FXO and FXS, and works fine from
several months.
Now I want to connect a device to TDMFXS that want a ring frequecy of
25 hz to activate: i am italian, and usually the ring freq is 20 hz.
The other time (I have used that device several times with other
asterisk installation) I have modified /etc//modprobe.conf and
2010 Oct 21
2
1 way audio asterisk 1.6
Hi
?
I ?wonder if?anyone could give some light on SIP NAT.
I've having a friken headache with SIP NAT 1 way audio.
Client - NAT? - NAT - Server
Client can hear users from server side
but server cant hear client.
?
Ive tried every possible settings
externip set
localip set
NAT= yes / route
directmedia yes/ no
?
Ive check the sip headers in the debug mode and its using the external address
in
2010 Jun 08
6
reloading realtime sip peers
Hello,
I noticed that changes to realtime sip peers are not applied until a
'reload'. A 'sip reload' does not make any changes to realtime sip peers.
When changing for instance the mailbox-parameter in the realtime
sip_buddies table, the change is not applied with a 'sip reload'.
For every change there is a complete 'reload' necessary.
Why does a 'sip
2010 Oct 28
5
being bombarded with SIP packets
Over the last two weeks, we have had at least two "incidents" where our
asterisk server got flooded (a hundred or more per second) by SIP
packets. Once from 114.31.50.10, second time from 173.212.200.146. We
became aware of the problem when bandwidth started suffering because
asterisk got very busy sending back replies or rejects (dunno which, I
didn't investigate it any further).
2010 Oct 16
6
Remote Unix Connection
Hi,
Does anyone know where this is suddenly coming from?
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Thanks
Dan
p.s. sorry about the last post. hit the mouse by mistake and it sent the email.
-------------- next part
2010 Oct 20
4
Recommendation for a new server
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm
2010 Nov 05
3
Elementary question - accessing feature codes from cell phone
Hi, please forgive me for this (hopefully) simple question. I cannot seem to
find an answer or solution while searching around.
I want to be able to call in to my server using my cell phone and be able to
set call forwarding for my extension and enter a phone number and also be
able to call in to that extension and disable the call forwarding. I see I
can do this through the ARI web interface