Well now would be a great time to come back, Doug! We miss you! 8)
--Rob
-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]
Sent: Monday, 26 June 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
Asterisk Users Mailing List -
(Try again from the proper email address)
--Rob
-Original Message-
From: Rob Thomas
Sent: Thursday, 22 June 2006 12:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] syntax error
That's freePBX or AMP code that we've since fixed
Third time's the charm.. (Email server is sending from wrong address!)
--Rob
-Original Message-
From: Rob Thomas
Sent: Thursday, 22 June 2006 12:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] syntax error
That's freePBX or AMP code
And I'll resend this one too. Silly scalix.
--Rob
-Original Message-
From: Rob Thomas
Sent: Thursday, 22 June 2006 12:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] zapata.conf: recent changes?
Looks like you've stopped compiling
: [Asterisk-Users] syntax error
From: Rob Thomas
That's freePBX or AMP code that we've since fixed - The
replacement line is
exten =
s,2,GotoIf($[${CALLERID(name):0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?4:3) ; check for old prefix
Yes, ok. I'm gradually fixing all the code
for it.
--Rob
(In a much better mood now 8)
- Original Message -
From: [EMAIL PROTECTED] on behalf of Brian Capouch
Sent: Thu, 6/22/2006 8:38am
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] syntax error
Rob Thomas wrote:
And once
Well, I wake up this morning, and aussievoip isn't up. I ring godaddy,
who _were_ hosting it, and they say that the machine's been compromised,
and you can't have your data. Nyah Nyah.
I spent 1 hour and 38 minutes on the phone to them, trying to convince
them to let me somehow get access to it,
The magic command you want is 'WaitExten' - 'show application waitexten'
on the asterisk command line. 'show applications' is also a good one.
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dinesh Nair
Sent: Sunday, 19 March
We buy them? Is this a trick question?
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ed Greenberg
Sent: Sunday, 19 February 2006 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
To quote Kevin:
DTMF handling in the trunk is in a state of flux right now. It won't be
resolved until this weekend.
Don't use SVN for a production system, it's lots broken right now. If
you really must, stick with r8786 for a while.
--Rob
-Original Message-
From: [EMAIL PROTECTED]
From the usual place, http://www.grandstream.com/BETATEST/GXP2000/
Note, there are two (and it took me a bit of a while to figure out)
images to be loaded. Copy the ...a.bin's and the .bin's to your http
provisioning directory, and reboot. The phone _must_ load the .bin
files before it
To check if it's the same problem, set your system clock back 2 weeks.
If it gets better, then the upgrade didn't take. If it doesn't get
better, it's something else.
--Rob
-Original Message-
Very greatful to find this I have upgraded to 1.2.3 but
still have no sip-sip audio!
When you did the make install
of asterisk, it gave you a whole pile of modules it didnt know about, some
of which no longer work with trunk. The easiest way to fix this:
# mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.old
then do a make install again
--Rob
The question is somewhat ludicrous, and Im
slightly surprised that no-one has sat down and done the maths about bandwidth utilization.
So I did.
To handle 5000 calls coming in over a PRI,
youd need 210 or so T1s or 170 E1s.
All of those would generate 320Mega BYTES of
data per second
Check http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom
From that article:
There is an 'allpage.agi' now available at
http://aussievoip.com.au/allpage.agi. Documentation is available in
I'm the author of that, and I've actually re-written it, because I was
pretty unhappy with the
After upgrading to 1.2.0 (from a three-week-prior CVS version), I've
suddenly had people starting to complain of lost calls. They'd be there,
and suddenly they'd drop out - they could be in a conversation, or more
often, the caller would be on old, and suddenly the light would go out,
and the
SHUT UP.
--Rob
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Look at zaptel/zconfig.h and see what is uncommented.
[...]
I am using the KB1 echo canceller.
The 'new' echo canceller is MG2. Please test. This is _extremely_ good
on PRI, but it's been reported to have some issues on 2 wire circuits.
--Rob
___
I've had an absoloutely fantastic run with the new KB1 patch currently
on mantis - http://bugs.digium.com/view.php?id=5520
The Digium guys are looking for feedback, please apply and test - If we
can get some positive feedback, it might make it into 1.2!
--Rob
All the unknown device means
is that your lspci doesnt know what the card is. Thats
all. Nothing more.
--Rob
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad
Sent: Tuesday, 4 October 2005 7:43
AM
To: asterisk-users@lists.digium.com
Subject:
/lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No
such device
This is correct. Your card uses 'wctdm' or 'wcfxs' depending on what
version of asterisk you're using.
--Rob
___
--Bandwidth and Colocation sponsored by Easynews.com --
I'm hosting soft-switch.org now - Steve has said he doesn't want FTP, so it's
all http now. Feel free to update the wiki.
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Thursday, 22 September 2005 11:50 AM
To: 'Asterisk Users
I use the Via PD-1 motherboards. They have 2 ethernet interfaces,
4 serial ports, 6 USB and a PCI slot. (And they have a fan 8)
Unfortunately, VIA can't supply the demand for these things at the
moment 8-(
http://www.viaembedded.com/product/epia_PD_spec.jsp?motherboardId=241
--Rob
It's a udev configuration. Read UDEV.txt, or, read the AMP wiki
http://aussievoip.com for a step-by-step.
--Rob
-Original Message-
From: [EMAIL PROTECTED] on behalf of Jachin Rupe
Sent: Wed 7/09/2005 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
-Original Message-
Thanks for the input Tony, but the instructions that Rob Thomas wrote
took care of my issue.
Thanks again to both of you!
You're welcome, Happy to help.
--Rob
___
--Bandwidth and Colocation sponsored by Easynews.com
You need to care about the _actual_ error, not the report there is an
error. The error is (usually) reported to the console. Reboot the
computer and type this:
dmesg -c /dev/null
modprobe ztdummy
dmesg
The output of the second dmesg will show you exactly what the error
message is.
Being that
-Original Message-
Of course your problem will become the fact that virtually no SIP
devices support more than 5 line buttons. The devices that do
support more than 5 line buttons don't run SIP. The only device
that I know of is the SNOM, but I've never used it. The Cisco
1: Upgrade your GXP to the latest firmware. See www.grandstream.com
2: [line1] number [send] speak [hold] [line2] number [send] speak
[transfer]
--Rob
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Sounds to me like bad RAM. Try running
memtest (your Fedora CD has it, just type memtest at the cd boot
prompt)
--Rob
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Wednesday, 6 July 2005 10:45
AM
To: Asterisk
Users Mailing List -
What's wrong with the standard 2.6 ztdummy?
It doesn't use RTC. I'm assuming you mean '1.0.8' as 'standard'.
How does HEAD zaptel interact with 1.0 asterisk?
Shouldn't cause any problems.
--Rob
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If you have Zaptel cards, does setting the build to USE_RTC use that
timing source in preference to the Zaptel card interrupts?
If you have a zaptel card, you shouldn't be loading ztdummy. So,
therefore, no problems!
--Rob
___
Asterisk-Users
Problem:
It seems the situation is improved when I remove the regsiter =
statements in my sip.conf.
Cause:
If your internet connection is down, your DNS isn't working. IF your DNS
isn't working, it won't be able to resolve names. If it can't resolve a
name, it will sit there trying until it
In my experience, several seconds of delay becomes apparent over time
when
using an internal clock source. Seems its a clocking/timer issue.
Yes. Meetme can have horrible issues with timing. This _has_ been fixed.
If you download the CVS Zaptel drivers, use a 2.6 kernel, you can use
RTC
The best thing for AMP support is to join the #amportal channel on
freenode.
irc.freenode.net
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Subject: [Asterisk-Users] Requirement for internal calls
I made it to configure some SIP Extensions, with the
etc. However, I missed the initial part of this thread. Why is this
an
issue? I went to voip-info.org today just to see what was going on
(while writing a googleapi tool to pull all cached docs from a given
domain) and it was running and appears to be there. Nothing on their
main page or
Also, I do not have RTC support in the kernel since the headers are
included from ztdummy, I thought that Tony said that it is not
required. Do I need RTC support compiled into the kernel?
I was going to reply to your first message, but then I thought I'd see
if you'd figured it out yourself.
output of zttest:
Opened pseudo zap interface, measuring accuracy...
99.938965% 99.951172% 99.938965% 99.963379% 99.963379% 99.951172%
99.951172%
Ah. Don't be scared. This is actually -correct- when using RTC. Let me
demonstrate:
[EMAIL PROTECTED] zaptel]# ./zttest -v
Opened pseudo zap
I would think the easiest way is to use an AGI..
exten = _001162.,Dial(Zap/g1/${EXTEN}/tT); (Anyone can call NZ)
exten = _0011.,Exec(checkperms);
checkperms.agi would then match against a list.
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Theres a less than sign missing it
should be:
patch -p1 ../asterisk-oh323-0.7.2-pre1/openh323_1.13.5-make.patch
(That will stop it hanging at least, not sure about anything
else with H323)
--Rob
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
I've finally got a chance to play with 1.0.1.9, and the wiki has been
updated. At the moment, I don't know of _any_ bugs with it. I'm yet to
play with complex things like early dial, and will update the wiki as I
find information.
http://aussievoip.com.au/wiki-GXP-2000
I've just been given
Note, there's a whole wiki dedicated to the GXP-2000 -
http://aussievoip.com.au/wiki-GXP-2000
Photos, reviews, buglists, firmware updates, everything. Go wild 8)
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Peter Svensson
Also for some odd reason when I ring an extension attached to my
sipura 2100 ATA it takes it about 12 seconds to start ringing after I
dial it (sits there with dead air on the calling phone).
After you dial, push '#' to actually start the call. Or update the dial
maps in the sipura. But # is
The unofficial GXP-2000 resource, bugs, and information page is at
http://www.aussievoip.com.au/wiki-GXP-2000
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Thursday, May 12, 2005 5:53 PM
To: 'Asterisk Users
I've looked into this. The important reasons as to 'why this shouldn't
happen' are:
Requires a Database - (bad for flash, also very large)
Needs apache + php (+30 odd mb)
A fair whack of perl modules (+10mb)
== Too large, too cumbersome.
--Rob
-Original Message-
From:
As no-one had actually put any technical details about how things work,
I wrote up a review of the GXP-2000 today.
http://www.gladstonewireless.net/tiki-index.php?page=GXP-2000
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
Well, no, it looks like the 8S661FX (which is actually called
8S661FXM-RZ or FXM_P_ for Socket 478) has 5v PCI slots.
http://www.giga-byte.com/Motherboard/FileList/ProductImage/photo_8s661fx
m_rz_big.jpg
and
http://www.giga-byte.com/Motherboard/FileList/ProductImage/photo_8s661fx
mp-rz_big.jpg
Is
-Original Message-
Is that what your motherboard looks like? Coz those are *definitely*
5v
slots.
And then I come to the slow realisation that the 10 is 3.3, and the 5 is
5. Sigh. Apologies.
--Rob
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On the subject of phones..
The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model)
phones can all do what he wants. ie, have multiple lines with blinking
red lights when a call arrives on that line.
How about phones that can indicate if an extension is busy or not - eg,
Busy Lamp
You want a _cheap_ reception phone? I don't think you are going to get
this.
Heh. I had a sneaking suspicion that was going to be the answer 8)
--Rob
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