Displaying 20 results from an estimated 1000 matches similar to: "Load Balancing over 2 E1 Lines"
2008 Mar 04
2
Problems configuring Astribank
Hi, all
My Asterisk uses a Digium TE120Pand I would like to add an Astribank
zaptel_hardware sees is, but I cannot get it working
pbx:~# zaptel_hardware
Argument "IRQ" isn't numeric in numeric comparison (<=>) at
/usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114.
usb:005/002 xpp_usb- e4e4:1131 Astribank-8/16 USB-firmware
pci:0000:04:00.0 wcte12xp+
2008 Jan 30
7
Problem with DTMF dialing
Hi all
I have a small problem here. I asked this question on another asterisk
mailing list, but nobody seemed to be able to help me there.
We are running
* Asterisk 1.4.17
* Libpri 1.4.3
* Zaptel 1.4.8
on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo
cancelation and a quad FXO card.
We have 4 analog lines, one of which is a Cellphone line for least cost
2008 Feb 26
6
[URGENT] Zap channels fail to load
I have spent some time this morning trying to add an Astribank to our
current Asterisk, but it failed, so I just removed the hardware,
restore the config files to the original setup and started asterisk.;
I could see that no Zap channels are started so I did load chan_zap.so:
pbx*CLI> module load chan_zap.so
[Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
Already have an
2008 Mar 05
4
{s} - extension
Dear all, I have small question
in sip.conf I added
[service]
type=friend
;username=
;secret=
qualify=900
host=X.X.X.X
dtmfmode = rfc2833
disallow=all
;allow=g729
allow=gsm
allow=alaw
allow=ulaw
and I can proccess incoming call from soft phone only I calling on
number that is used in extensions.conf(in example below it is 1)
exten => 1,1,Answer;
exten => 1,2,Playback(hello-world,skip);
2007 Aug 17
3
Lock extension from asterisk
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi all
I am working in a new set up with Grandstream GXP-2000 handsets. I
like those phone, but they lack a feature I need: the phone cannot be
locked by the user.
What I actually want is a user to be able to avoid someone else making
calls from his phone without giving him access to SIP configuration
access to the phone.
i.e. let say I want user
2007 Aug 18
1
Best way to detect unknown and/or private incoming caller-id?
I am aware of how to match a particular caller-id or a caller-id
pattern and do something with the call like this:
exten => 15554441212/_888NXXXXXX,n,Playback(GoAway)
What I am curious about, is the best way to block unknown, private and
000-000-0000 calls.
I know I can do this for 000-000-0000 calls:
exten => 15554441212/0000000000,n,Playback(GoAway)
Is there a better way to catch
2008 Feb 22
5
load balancing SIP extensions
What I would like to do is have two identical *
servers which accept registrations of sip extensions
4000-4999.
If I define a rrDNS or LinuxHA then I should have
load-balanced registrations.
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001 tries to call 4002 then I
would like to do something like:
- lookup 4002 on *1, try to establish a call if it's
2008 Mar 01
4
Cisco 79xx users/consultants, 7970G color in particular share information
I would like to get in contact with users/consultants who are or have
worked with the Cisco phones and Asterisk to trade information.
Cisco has reluctantly made SIP available on their phones and most of the
information on voip-info and other wiki's appears to be reverse
engineered. There is a wealth of information out there which is
terrific.
I have a client with about 40 phones
2008 Feb 27
3
About faxes recived through a PRI and passed to a fax machine connected to a FXS port
Hi, all
I want to configure a few FXS ports in an Antribank-16 to be able to
receive faxes sent throught a PRI:
E1 ==>Zap * ==>FXS * ==>Fax machine
My asterisk box has a Digium TE120P (for the PRI).
Versions are *=> 1.4.17 | Zaptel=>1.4.8 | libpri=>1.4.5
The Astribank is not configured yet, because I am a little bit
confused about how to do it.
Let's say I configure
2007 Dec 09
1
Installing/configuring TE120P debian way
Hi all
I use asterisk (1.2 brach) from debian official packages and it works fine.
Now I need to install and configure a Digium TE120P card, but I cannot
find any guide to install it using debian packages.
I would like to know if anyone of you knows about packages that would
include the necessary kernel modules or any other method that won't be
broken when the asterisk packages are updated.
2004 Jul 10
2
New Asterisk bounty: SIP simultaneous registry
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry
From the WIKI:
Contributions
Manager: Daniel Jimenez (cuban)
Bounty: $50 USD
Date opened: July 10, 2004
Contributors: cuban ($50)
Detail
Yes, Yes I know you could do all sorts of fun with the dialplan to
produce a similar effect, but I still would like to be able to do this.
Plus it's easy money :).
I
2004 Jun 08
4
AS5300 and Asterisk
Hey all,
I have an as5300 I use for dial in customers, we have 4 PRIs on it.
We have a few free channels on it. I'm wondering if I setup SIP on the
as5300 I can have asterisk use the free channels for dial out.
I'd still have to use my TDM04B for incoming calls, but at least I can
expand my outgoing.
Anyone done anything like this before? I've never messed with VoIP on
Cisco
2015 Nov 15
21
[Bug 92961] New: Xorg freezes (only mouse and ssh are still working)
https://bugs.freedesktop.org/show_bug.cgi?id=92961
Bug ID: 92961
Summary: Xorg freezes (only mouse and ssh are still working)
Product: xorg
Version: unspecified
Hardware: x86-64 (AMD64)
OS: Linux (All)
Status: NEW
Severity: critical
Priority: high
Component: Driver/nouveau
2004 Jun 06
2
BRI In the states
Hi all.
I've ordered a TDM400P with 4 FXO, but after using my X100P I'm thinking
about returning the TDM400P because of bad echo issues. If I do get the
echo issues I'll look at digital options.
My question: Is anyone using ISDN (BRI) in the states? I've heard
ISDN4LINUX devices suffer bad echo but chan_capi works great. All the
chan_capi cards I find though are for overseas
2005 May 12
2
Problem with Polycom SP 500 and Cisco PIX
Hi everyone,
I'm very new to all this, so please forgive me if I have the terminology
mixed-up.
We are preparing to install an Asterisk IP PBX over the weekend and I have
an issue with the Polycom SP 500 phones we are trying to use. My problem is
regarding DHCP.
Our DHCP server is our Cisco PIX 501 firewall. I've specified option 66 and
the phones connect to the FTP server
2005 May 16
2
Winbind problem when exec freeradius
Hil list!
I'm trying to authenticate Active Directory Users via freeradius. I
can do it in a general case (user and domain) without
problem. Now I have to do it restricting the authentication to the
members of a group.
I can exect the script (as is put in radiusd.conf) correct from the
command line:
Deb:~# /usr/bin/ntlm_auth --username=javi2
--require-membership-of='AAMM\MyGroup'
2004 Jul 06
3
Dialing out of a voicemail message?
Anyway to make hitting `0` during a voice mail dial an extension? The
bosses used to have that feature and love it.
Their VM prompt would say: "Hello, My name is blah blah. I am currently
unavailable. If you would like to speak to an operator press 0 now,
otherwise leave me a message".
Extension 0 exists, but dialing it during a VM prompt does nothing.
Thanks,
--
Daniel Jimenez
2004 Sep 03
1
BIG ISSUE with SIP, not sure where to go but it's killing asterisk.
I frequently get this error message, it repeats itself hundred/thousands
of times and never stops.
chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again...
During this period, I can make no SIP calls what-so-ever. The only way
I've been able to stop it is to killall -9 asterisk. Doing a restart now
doesn't respond.
Anyone know why?
--
Daniel Jimenez
2017 Feb 14
6
Registration of native routines
Registration of 'native routines' (entry points in compiled code loaded
into R) has been available for over 14 years, but few packages make use
of it (less than 10% of those on CRAN with compiled code).
Registration has similar benefits to name spaces in R code:
- it ensures that the routines used by .C, .Call etc are those in your
package (without needing a PACKAGE argument).
- it
2006 Nov 24
2
Card don't hangup but Asterisk hangup
Hi ,
I have a problem with a X100, i do a external call to the asterisk
server . The dialplan its simple answer and hangup..
when it's done , the telephone which i did the call , is in line but
asterisk server is finish.
I'll apreciate all your suggestion. Greetings, txus.
The asterisk output:
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn