Displaying 20 results from an estimated 2000 matches similar to: "Dual PRI fail over"
2003 May 22
3
nfas on T400P?
Can T400P be configured for nfas (one d-channel providing signaling for more
than one span)?
Thank you.
Alex Zarubin
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2004 Nov 05
3
Resources for optimizing code
I want to eliminate certain observations in a large dataframe (21000x100).
I have written code which does this using a binary vector (0=delete obs,
1=keep), but it uses for loops, and so it's slow and in the extreme it
causes R to hang for indefinite time periods.
I'm looking for one of two things:
1. A document which discusses how to avoid for loops and situations in
which it's
2004 May 14
7
What's in ${EXTEN} ? Why does voicemail prompt for an extension?
Why does voicemail prompt me for an extension instead of just asking my
password?
[voice-mail]
exten => 99,1,VoicemailMain(${EXTEN}@inside)
exten => 99,2,Hangup
Paul Mahler
pmahler@signate.com <mailto:pmahler@signate.com>
<http://www.signate.com/>
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
2004 Mar 17
4
can't logon to voice mail - bad password
I have one SIP extension that can't logon to voicemail. The log file says
-- Incorrect password '3213' for user '4035' (context=other)
even though the context in voicemail.cnf says
4035 => 3213,Bill Smith
Thanks!
Paul Mahler
mail:pmahler@signate.com
phone: 650.207.9855
fax: 877.408.0105
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2004 May 14
4
How to Echo extension number to caller?
I need to dial an extension that tells me what extension I'm dialing from.
I'm running a bunch of analog phones off a channel bank to * over a T1. I
have the following in extensions.conf.
exten => 98,1,SayDigits(${EXTEN})
This says the digits the caller enters on the keypad, not the extension they
are calling from.
Thanks Guys!!!!!!!!
Paul
Paul Mahler
pmahler@signate.com
2003 Dec 18
2
Cisco 7960 - can't traverse NAT?
Might be a stupid question, but is there a default gateway set on the 7960?
-----Original Message-----
From: Paul Mahler [mailto:pmahler@signate.com]
Sent: Thursday, December 18, 2003 7:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT?
I have a 7960 running behind a firewall running NAT. From a telnet session
to the 7960, I can't ping
2010 Jun 12
2
Qwest PRIs
Hi,
I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm
using an OpenVox D410E and the drivers are loaded. My system.conf looks
like this:
# Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED
span=1,2,0,esf,b8zs
bchan=1-24
# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
These
2004 Nov 19
5
Asterisk and H.323 Gatekeeper
Hello,
I am new to this list and to asterisk and going through the archive file I
did not find an answer to my problem.
I have a VoIP network working fine with multiple gateways registered to a
Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in
that network and also successfully registered two X-Lite SIP Client to
asterisk that call to each other.
I want to connect to
2004 Jan 30
3
How do you turn on the 7960 msg waiting light?
Does anyone in Asterisk land know how to turn on the message light on the
back of the earpiece of a cisco 7960 when a message is waiting?
Thanks!
Paul
Paul Mahler
mail:pmahler@signate.com
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2004 May 10
1
Terrible TICKING sound
i'm getting a tick every second or so on all my calls. All channels are zap
channels.
Does anyone know how to fix this?
Thanks!
Paul
Paul Mahler
pmahler@signate.com <mailto:pmahler@signate.com>
<http://www.signate.com/>
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
2004 May 02
1
Why don't I get a ringing sound?
I am using the following macro to dial a ZAP channel. When I dial in, *
answers and I go to voicemail. I never hear any ringing, though. It doesn't
work with the Ringing command before or after the Dial command.
[macro-zapdial]
;
; call a ZAP extension for ${ARG2} seconds, and then voice mail
; ${ARG1} - Extension
; ${ARG2} - Time to ring
exten => s,1,Dial(ZAP/${ARG1},${ARG2})
exten
2006 Jan 22
5
T3 Mux and Asterisk Question
I have a T3 coming from my carrier. From there I want to use an Adtran
mx2800 T1 Mux to break the T3 into 28 T1/PRI which feed into seven quad
T1/PRI equipped servers.
Everything seems very straight forward with the exception of the D
channels for the T1/PRI.
I am not very familiar with large circuits such as T3s. I know that I
can use one D channel per set of quad port on each server. So if
2004 May 24
3
100 analog phones?? HOWTO?
Does anyone know the best approach to take for handling 100 analog
phones? It seems to me that a chassis like Carrier Access or Adtran
would work. The chassis would do much of the hard work of converting
the analog sound to data.
Any recommendations on hardware for the chassis?
...Jeff
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
2004 Jul 04
4
Asterisk Book
If anyone is interested in getting a book on asterisk I would recommend
checking out http://www.saww.net/asterisk/
2004 Jun 22
5
CISCO 7960 Goes missing
I've got a number (10) Cisco 7960's connected to my network. All the phones
work perfectly except one.
The asterisk console keeps throwing up:
Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer
'4001' is now UNREACHABLE!
Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer
'4001' is now REACHABLE!
Jun 22 15:42:08
2006 Apr 28
3
Dual Timing Sources
Hi,
With a digium dual PRI card (dual span). Is there any reason I can't
have both PRIs being PRIMARY timing sources? They are both from
different CLECs, and as such I need them both to do their own timing.
2005 Mar 22
2
Asterisk locking up - 99.9% CPU
Hello
We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work
with our call agent.
Unfortunately **VERY** frequently, asterisk stops responding and goes to 99.9%
CPU. There is no debug output or other information that indicates there is a
problem...
Rather than continually restarting, can anyone make suggestions as to how we
can track this down **OR** has anyone got the
2006 Feb 02
2
RE: 5, 000 concurrent calls system rolloutquestion
I don't think they are doing it with one Asterisk box. They did say "one rack of servers". Well, that might mean up to 50 computers if they are using blade servers.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Todd
Sent: Thursday, February 02, 2006 10:21 PM
To:
2006 Jan 09
1
Second edition of my * book has been released
How does it compare with the O'Rielly book?
Does it include information on CVS, or primarily on stable?
Can it be provided to customers, or is it more sysadmin oriented?
Regards,
Greg
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul
Mahler
Sent: Thursday, January 05, 2006 9:45 AM
To: 'Asterisk