Displaying 20 results from an estimated 900 matches similar to: "Asterisk locking up - 99.9% CPU"
2005 Feb 04
2
AU caller ID with Sipura SPA-3000
Hi All,
I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and
out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN
Line" tab) so Asterisk answers the call rather than the SPA-3000. It
is all working perfectly except I can't get the SPA-3000 to pass
caller ID to Asterisk. It passes "Display Name", "User ID" and any
"PSTN
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 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
2005 May 18
2
FREE music for downloading
Need new Music on Hold for your PBX?
Signate is happy to make a variety of classical music selections available,
sampled at rates that are appropriate for telephony. There is no charge.
The selections feature Elena Kuschnerova, pianist, and Lev Guelbard, violinist,
playing public domain pieces that will give callers a classic impression of you
or your company . Click on the link to see a list
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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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
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 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
2006 Jan 10
1
Still an open Seat in London for Next Weeks Signate intro to Asterisk Course
We still have a seat open in our Asterisk training course next week in
London. You can find more information at our Web site, www.signate.com
I'm going to be teaching the class.
Paul
2006 Jan 10
3
The second edition of my Asterisk book is now available
The second edition of my book "VoIP Telephony with Asterisk" is now in
print and available. You can find out more about it at our web site
http://www.signate.com/products.php
This book is written for beginners. It will make it easier for you to get
started. The second edition is reorganized and expanded.
Thanks,
Paul
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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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:
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
2005 Mar 25
1
We just released our new Asterisk Installation CD set. with 24/7 monitoring
Here's our recent announcement of our new Asterisk Installation CD set:
Signate has announced its new Asterisk Installation 2005 CD Set. It's, a
complete software PBX (private branch exchange) telephony appliance in a single
package. The CD set installs Linux pre-configured for telephony, a stable 1.0x
distribution of the open source Asterisk PBX, and Signate's optional, free PBX
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
2019 Jul 27
2
Help on Optimization Remarks
Dear llvm-dev community,
I am trying to analyze the optimization remarks generated through clang
using -fsave-optimization-remark with -O3.
--- !Analysis
Pass: loop-vectorize
Name: CFGNotUnderstood
DebugLoc: { File: c-ray-mt.c, Line: 177, Column: 2 }
Function: main
Args:
- String: 'loop not vectorized: '
- String: loop control flow is not understood by vectorizer
I tried to look for
2004 Jul 08
8
FINALLY! a good book about Asterisk.
There is finally an introductory book about Asterisk!
It looks like Paul Mahler at www.signate.com wrote it
with a lot of help from Digium. I looked at the sample
pages, it looks great.
__________________________________
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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
2005 Aug 29
1
Moving to New Zealand
Is there anyon here currently in New Zealand that use asterisk, I need
to help getting voice and internet services. I will be moving in a week.
Any help would be great. Please use the details below to get ahold of
me.
Thanks in advance.
James Jones
Signate, LLC
james.jones@signate.com
415.442.4012 (office)
413.771.1402 (office)
413.977.6482 (mobile)
413.667.3105 (fax)
665 Third Street
Suite
2004 Jun 07
1
Seeking Volunteers for an Intro to Asterisk Course
Our company, Signate, is going to be offering a three day introductory
Asterisk training course, the first of a series.
The first class will be in San Francisco the week of June 28. It will be a
beta test to get the kinks out and we will not charge for the class or the
materials. Anyone who attends is responsible for their own travel and
lodging, if necessary.
I doubt anyone reading the lsit