Displaying 20 results from an estimated 10000 matches similar to: "8 FXS in Asterisk Server"
2005 Aug 11
8
Newbie Question: Building an Asterisk system to replace an old PBX but using existing phone
I have a brief from a local hotel to build a PBX using Asterisk but they
want to use their exisiting telephones and wiring from an old PBX that
no longer works.
Basically, I can build the system but an looking for a card that will
allow for upto 20 extensions to be wired into the back of the PC. Doeas
anyone know of a solution to this
Sean--
ICQ: 679813 FidoNet: 2:263/950
Jabber:
2005 Aug 14
2
Problem with FWD connection rejected
Using the install instructions for A@H, I setup a FWD account, this I
tested using X-Lite and it works okay,
Nowever I cannot make calls to fwd using Asterisk, my log showes:
Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected:
Registration Refused
Aug 14 21:06:59 NOTICE[1324]: Registration of '689482' rejected:
Registration Refused
According the to info screen of
2005 Aug 13
3
One more newbie question
Ok, I am going for A@H with the CentOS iso disk. Installed and just
checking a few things out.
My other question is this, which I forgot to ask before. We have no
Broadband here and more than likely will never have, so I am just
looking at building Asterisk to handle inbound and outbound calls, at
home via a ISDN card and for my hotel job via a PSTN line setup. Is this
very complicated to setup
2005 Aug 18
3
Preventing an extension from dialing certain outbound codes
Is there anyway to prevent an extension from dialing certain codes. ie I
want to prevent extension 203 from dialing number which start with 00
087 086 etc
Sean
--
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|VOIP: FreeWorldDial 689482 VOIPBuster thecivvie |
|GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc |
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2005 Sep 14
1
Indications for Ireland
Hello asterisk-users,
Just curious if anyone has the indications for Ireland, tried
googling for it to no avail.
Sean
--
+---------------------------------------------------+
|VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie |
|GPG Key http://thecivvie.fastmail.fm/thecivvie.asc |
+---------------------------------------------------+
Strange things happen under the midnight sun
when Men
2005 Aug 19
1
Console OSS or ALSA for 3.4
What do I need to install for OSS or ALSA for console use only
Sean
--
+----------------------------------------------------+
|VOIP: FreeWorldDial 689482 VOIPBuster thecivvie |
|GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc |
+----------------------------------------------------+
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Name: smime.p7s
Type:
2004 Aug 10
11
CAPI call transfer
Hi,
I am having trouble configuring CAPI so that call transfers work.
I make a SIP call to asterisk which goes out on ISDn via CAPI. Then
I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.
The error message given by asterisk is that it chan_capi can't find
an
entry for the outgoing msn for the transfer however the outgoing msn
is the
2005 Jul 03
2
play message to callee beforeconnecttoincomingcall
Thanks for the suggestion, C F, but the problem is there is a rather big
database application behind with many users, so a static configuration
is not suitable for my needs. i am working mostly with realtime and agi.
regards,
roland
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F
Sent: Sunday, July 03,
2004 Sep 05
4
Asterisk & sudo from httpd
Hello!
I want to use "asterisk -rx "show version"" from a php script called in
the browser using the local apache, which runs as user "apache".
Asterisk is running as root.
I added the following line to /etc/sudoers using visudo:
apache ALL = NOPASSWD: /usr/sbin/asterisk
When i am on the command line of my linux box it looks like this:
2004 Sep 03
2
Using AVM Fritz!PCI as zap interface
Hello!
Is there a way to use AVM Fritz!PCI as a ZAP interface and have it
configured for ZAP channels?
Thanx in advance!
Roland Zagler
mailto:r.zagler@fog.at
@fog smart partners
2005 Jul 12
3
Cisco 7940/7960 interdigit timeout
Hello list,
does anyone know how to change the "interdigit timeout" when using Cisco
IP Phone 7940/7960 with SIP-Firmware and Asterisk?
it's default value is 15 sec. but i have nothing found to set this in
tftp-config file etc.
Thanks in advance,
Roland
2004 Aug 18
1
Asterisk as SMS Service Center
Hello!
Is it possible to run Asterisk as a SMS Service Center and does it work
over a directly connected ISDN (CAPI) interface card?
Did anyone already use Asterisk for that?
Roland Zagler
mailto:r.zagler@fog.at
@fog smart partners
2006 Jun 22
1
PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM
Hi to all,
we are searching for a hardware based DSP solution for use
with Asterisk based on PCI or MiniPCI to reduce main processor
load and to use embedded boards with Digium E1/T1 cards like
TE410P.
does anyone know about any manufactorer of those cards or someone
who is able to develop/build such cards?
Specifications:
PCI or MiniPCI
up to 120 concurrent transcodings
Codecs: G.729/G.729A or
2009 Jun 01
3
[Atcom] Asterisk + LAMP on 128MB RAM?
Hello
I'm thinking of selling an Asterisk server based on Atcom's IP02
solid-state unit with one FXO and one FXS ports:
http://atcom.cn/En_products_IP02.htm
By default, this unit based on a 400MHz Blackfin 532 chip only has
64MB RAM and 256MB of NAND flash. Those can be increased to 128MB and
1GB, respectively.
Do you think I can install Linux + Asterisk + LAMP (replacing MySQL
with
2004 Nov 15
3
Memory Consumption
Hello,
I use Asterisk 1.0.2 on a RedHat Enterprise Server 3.0 (Kernel 2.4.21)
and i experienced that the memory consumption of the asterisk-process
started by the init.d-script raises continously. Now, after 3 hours of
operation (on our testing-system we have 30 concurrent connections to
another asterisk box using IAX2 and GSM codec) there is already 66MB
allocated. I think this could be ok, but
2005 Aug 01
4
IAX Devices Recommendation
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Does anyone have any recommendations on an IAX Desktop Telephone or ATA
Device. I currently have 2 of the SIPURA-841's on my local network and
now I am wanting to try an IAX Device at my remote office since I think
that it would be easier to configure through various routers than a SIP
Device. I just started to look at the Digium IAXy Single FXS
2005 Jul 21
1
SOLVED: TE410P card in an HP-Compaq DL380 G4 server
Hi to all out there using HP DL380 G4 servers,
i found a way to get the Digium TE410P with older firmware running on a
HP-Compaq DL380 G4 Server! Here's the step-by-step description:
1. download the latest BIOS (in my case it was 4.04 from date:
06/02/2005) for
the HP-Compaq DL380 G4 using the
"Systems ROMPaq Firmware Upgrade Diskette for HP ProLiant DL480 G4 (P51)
Servers"
Link:
2005 Jul 02
1
play message to callee before connect toincoming call
Thank you, Robert!
The announcementfile plays well, but at Dial-option "m" i have to
specify a MoH class,
that is something i cannot use (as i wrote in my post).
Background command waits for a user input, but the caller should be
connected to
SIP Phone 100 after it has answered and the announcement has been
played. Before
connecting to SIP Phone 100 the caller should hear a
2008 Jun 20
1
FXS port doesn't provide dialtone
Hello everyone,
I want to connect a fax to an FXS port (TDM420P). For testing purposes,
I connected an analogue phone to it first. However, when I pick it up, I
cannot hear anything at all.
The power cable is plugged into the card, the port is configured to use
fxo-signalling. Also, immediate=no. Here's the files:
/etc/zaptel.conf:
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand
2004 Aug 22
1
Spandsp - opencall.org offline
Please can someone send me the .tar.gz file of spandsp, the site is
offline and i didn't find it anywhere!
Thanxxxx!
Roland Zagler
mailto:r.zagler@fog.at
@fog smart partners