Displaying 20 results from an estimated 1000 matches similar to: "Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?"
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
2009 Aug 30
1
Help me testing this webphone at www.VisionVoIP.com
Greetings everyone,
I've been trying to make this java based webphone work for everybody
visiting my website, but seems like for many users it doesn't work. In order
to get a better idea what is the success rate of this webphone, I would
appreciate help from anybody who could make a few calls from it within North
America and if it doesn't work, send me what error you get, or if it
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)
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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
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 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
2011 Sep 18
2
DTMF problem
[This email is either empty or too large to be displayed at this time]
2010 Mar 18
6
Asterisk DIES with no trace. PLEASE
Thanks Zeeshan,
SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told me to reinstall asterisk again
But, i still having doubts about the problem :(
Thanks in advance
>
> Message: 10
> Date: Thu, 18 Mar 2010 00:21:06 -0400
> From: Zeeshan Zakaria <zishanov at gmail.com>
> Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE
>
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
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2008 Oct 17
5
How to add contexts in asterisk realtime?
Hi everybody,
How can we add new contexts in asterisk realtime module? All I could figure
out after googling is that a new context HAS to be declared in
extensions.conf with 'switch => Realtime/@<databasetable>' under the context
name declaration. This works fine as long as we are adding extensions only
to this one context, but doesn't give the freedom to add new contexts for
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 Oct 29
1
BLF in Asterisk 1.4.*
Hello everybody,
does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm
particularly interested in Asterisk 1.4.25.
Thanks in advance!
Phil
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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
2009 Jul 11
2
Suggestions for web based soft phones
For a while now I've been looking for a good web based soft phone solution,
but so far no luck. A few solutions which I've tried, both Java based and
Flash based, either don't work, or had bad sound quality. I need something
which I could put on my productions server for my clients.
Seems like good web based solutions are all paid ones, nobody is giving it
for free. Any ideas,
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 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
2009 Jul 14
3
Why CDR is recording dst value = h?
For a new project, I have written a dialplan and it is pretty straight
forward: The [dialout] context dials out a number, and h extension in this
context writes the CDR. But what is happening is that if the callee hangs up
first, all values in the CDR are fine, but if the caller hangs up first, the
'dst' column is always 'h'. I put a NoOp right in the begining of this macro
to
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 15
4
can't seem to register, status unmonitored
Hi everybody,
I am trying to register my softphone(twinkle) on an asterisk server.
Everything seems to be fine.
Here is the output on show registrations in twinkle:
Tue 18:57:51
nikhil: you have the following registrations
<sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013
208 is ip of the asterisk server.
on the server on doing 'sip show peers' , it
2007 Jul 18
5
In Vancouver is it a local to call from 778 to 604, and vice versa?
I've got a 778 DID for vancouver, but don't know if it will be a local call
if dialed 604 and vice versa.
What are the different area codes in Vancouver and why its easier to get 778
DID than 604?
--
Zeeshan A Zakaria
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