I'll assume you mean a Dell PowerEdge 2950. Sangoma's web
site says the
cards dimensions are 55mm(H) x 290mm(L). A Full-Length PCI card is
107mm(H) x 312mm(L). According to the PowerEdge 2950 Getting Started
Guide Page 10:
Left riser
PCI-X option: two full-height, full-length
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: Wednesday, August 15, 2007 7:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] why is
You have on your hands a broken UA, since it is not responding to the
changing nonce value.
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: Wednesday, August 15, 2007 7:52 AM
To: Asterisk Users
What we really need is for someone to pay Allison and get the lyrics
recorded in her voice. ;)
BTW, you just wasted about 30 minutes of my time while I looked around
that site at the versions written in languages I've used over the years.
:)
- Brad
-Original Message-
From: [EMAIL
Horseshit. Prior art is trivial. How old is Hylafax?
Cheers,
-- jra
It's never trivial if you're a small company. J2 has already won
settlements from several smaller companies, which gives it
precedence.
Once precedence is established, it's almost a done deal for future
On Wed, Aug 29, 2007 at 09:48:18PM -0400, Matt wrote:
I guess that's my point. I realize asterisk is open source
and FREE,
however, I wouldn't expect a commercial application to
crash as often
as I've seen asterisk go down.
Windows 98.
wouldn't expect != haven't
On a side note, does anyone have the URL to the AEL example so I can
write out an extensions.conf version for the wiki?
- --
Kind Regards,
Matt Riddell
Director
It's called queues-with-callback-members.txt in the /docs directory in
the source tree.
- Brad
Dozens of Dell PE2950s, mostly dual Xeon 5150s with 4GB RAM and two 73GB
drives. Some have TE412Ps and some have TE420Bs.
Also, 14 PE2850s (dual 3.0GHz, 4GB RAM, dual 73GB drives) with a mix of
TE411Ps and TE412Ps.
___
--Bandwidth and Colocation
I had this same issue with 601s, and I was able to fix it by defining:
progressinband=yes in sip.conf.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio
Sent: Tuesday, January 31, 2006 11:20 AM
To:
Title: Message
It
looks like the outbound caller ID is not being set properly. Most of the
carriers that I've dealt with will act exactly as you said if you do not set it
to what is expected at the 911 center.
In
particular:
Calling Number (len= 8) [ Ext: 0
TON: Subscriber Number (4)
For your second question, how about the application MailboxExists? You
could write a quick front-end that asked for the mailbox and then used
MailboxExists to test. If it doesn't, perhaps increment a counter (so you
can disconnect later if it exceeds some value) and then return to the
original
While I've never actually tried exactly what you're doing below
(constructing a variable name from strings and other variables), it looks
like the variable substitution you're attempting is not being done properly.
Try something like:
exten = s,3,GotoIf($[ ${NUM${mainLoop}_CMD} = Dial ]?5:7)
It must be microseconds that is being quoted, as even the 2626 that you
mention lists a less than 13.3 microsecond latency.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ankers
Sent: Thursday, February 23, 2006 6:54 PM
To: 'Asterisk Users
Perhaps in a similar thread, is it possible to somehow SET the state
of a hint from the dialplan? Perhaps a bit like:
Set(${ChanIsAvail(hint,234)}=Busy)
or perhaps have a pseudo-device facility where you can add
it to the
end of the hint list to hint-the-hint. Something
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Hackensack
Sent: Monday, March 24, 2008 5:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Passing variables over IAX2 --
IAXVAR patch?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
Sent: Thursday, March 27, 2008 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about PCI Slots for
DIGIUMs Boards
No actually
I actually just ordered 50 licenses to give this and the other
applications a try. I'll post my results to the list once I get them
and have had a chance to play around.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of faraz
Sent:
Russell Bryant wrote:
This is a slightly different approach, but have you seen the
state interface
code that is in Asterisk 1.6? There is a backport of the
code for 1.4 floating
around somewhere, I think. It allows you to specify a
different device for a
queue member that app_queue
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Lee, John (Sydney)
Sent: Thursday, July 31, 2008 3:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie in China: Red alaram in
Zaptel for E1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Sessions
Sent: Thursday, August 28, 2008 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Tips and Tricks: Dynamic
Subroutines inAGI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tilghman Lesher
Sent: Wednesday, September 10, 2008 5:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Write Asterisk CDR MySQL
records to
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian J. Murrell
Sent: Friday, September 26, 2008 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip forking needed for ekiga 3.0
I've read
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: Sunday, April 29, 2007 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Poor man's High Availability solution
Who resells
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
JR Richardson
Sent: Friday, May 25, 2007 11:31 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom or Linksys phones bootp
tftp config setup
Hi All,
Has anyone gotten the
Thanks Stefan! I was just thinking the other day that it would be great
if I could whiteboard in Spark.
Back on topic, I'm definitely interested in this web conferencing app.
I'll have to check it out once a .war is made available and I have a few
spare moments.
- Brad
-Original
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ricardo Carvalho
Sent: Friday, June 01, 2007 6:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] how can qualify=yes trigger some
external event?
Hi
qualify=yes generates events that can be viewed from AMI, they are:
'Event: PeerStatus'
'PeerStatus: Lagged'
'Event: PeerStatus'
'PeerStatus: Reachable'
The other fields give the peer name and like, for more
details view the chan_sip.c source, the calls you are
interested in
Please post the relevant portions of your sip.conf and extensions.conf
I'll bet dollars to donuts you have the same context defined as both
your regcontext and as a context in extensions.conf (or an .ael, or
whatever).
- Brad
-Original Message-
From: [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
John Treble
Sent: Thursday, June 07, 2007 10:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: [EMAIL PROTECTED]
Subject: RE: [asterisk-users] PRI Partial Re-Rounting
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Watkins,
Bradley
Sent: Thursday, June 07, 2007 3:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Reload in 1.4 clears regexten
Please post the relevant portions of your sip.conf and extensions.conf
UltraMonkey (www.ultramonkey.com) and MySQL Cluster
(http://dev.mysql.com/doc/refman/5.1/en/mysql-cluster.html)
It works a charm.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Justin Moore
Sent: Friday, June 08, 2007 2:13 PM
To: [EMAIL
Today, buying extra ports for stations having extra
bandwidth requirements
is acceptable as 10/100 LAN access is the norm.
But it could be painful to explain executives, every IP
Phone you bought
during 2007 will not keep up with 1GE LAN.
There is one other issue - I don't think
What does the output of 'show dialplan start' look like?
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Douglas Garstang
Sent: Tuesday, June 19, 2007 3:20 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Ex-Girlfriend
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Khaled Chehab
Sent: Thursday, June 21, 2007 7:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql
Yes mysql
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Thursday, June 28, 2007 3:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] network routing
This allows me to edit the IP Address of
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Fromm
Sent: Thursday, January 18, 2007 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue and Interface time out
I guess I'm missing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Fromm
Sent: Friday, January 19, 2007 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue and Interface time out
Does anyone have
What it actually does is tell the SIP channel driver to track whether or not
any given peer has a call to it. It can then subsequently inform the Queue
application so that another call will not be given to that user. If you did
not have the ringinuse=no in your queue definition, you would
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gordon Henderson
Sent: Friday, February 09, 2007 9:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Conferencing Phones ...
Anyone got any experiences
You have the PRIs set up to recover clock from the Asterisk box, is that
what you want? If so, you certainly do *not* want span=1,1,0 or 2,2,0
since that will make Asterisk think the 81C should be clock master. Are
there any telco-timed PRIs somewhere? If so, set up the PRIs on the 81C
to be
Are you saying that the Nortel will not allow you to set the clock to internal?
If so that's unfortunate, as it's the only reliable solution for you in this
situation. You really need your clock hierarchy to start at the received clock
from the telco.
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Bishop
Sent: Tuesday, February 20, 2007 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Passing a variable from one Asterisk box to
another
Allow me to register my interest in any and all things that tie Asterisk
information to Cacti. We use that here, and it's been on my to-do list
for a lgg time.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brandon Kruse
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Atis Lezdins
Sent: Tuesday, December 02, 2008 1:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, December 04, 2008 7:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] polycom no menu
Was messing with a polycom 501 and changed the IP from
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Tilghman Lesher
Sent: Tuesday, December 16, 2008 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.6
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Atis Lezdins
Sent: Tuesday, December 16, 2008 5:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.6
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent
Davidson
Sent: Wednesday, December 31, 2008 1:03 PM
To: m...@digium.com; Asterisk Users Mailing List -
Non-Commercial Discussion
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Klaus Darilion
Sent: Thursday, January 08, 2009 8:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AEL and
I just want to confirm but it seems that if-then-else is not permitted
in case structure.
It was not really documented but it seems to be the case.
Can anyone confirm?
No, if-then-else works fine inside a case statement. See inline
comments.
switch(${DIALSTATUS})
{
case
The method OpenAIS uses to communicate between nodes is
designed for a
very low latency local connection; it is not designed to work across
routed connections. Russell Bryant has spent some time
talking to the
OpenAIS developers about this, but so far there doesn't seem to be a
good
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, May 11, 2009 3:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hello, all. The little bit of reading I've done on lua makes me eager
to give it a try. However, when I try to install it (Asterisk 1.6.1.1
on CentOS 5.3), it is not available in menuselect. I have
installed lua
and lua-devel. I've seen very little about it in my Internet
searches.
Wow! Definitely a non-trivial patch. Alas, it does not work but the
errors are different:
[compu...@pbx01 asterisk-1.6.1.1]$ grep -i lua config.log
configure:42697: checking for luaL_newstate in -llua5.1
configure:42732: gcc -o conftest -g -O2 conftest.c -llua5.15
/usr/bin/ld:
My guess is that when running the compile test ( This line:
'configure:42995: gcc -o conftest -g -O2 conftest.c
-llua-5.15'
) it is necessary to add '-lm' in order to link in the standard math
library.
- Brad
One more bit of magic necessary here, as pbx/pbx_lua.c has includes
That worked. The system is still in enough of a test phase
that I can
destroy it again and rebuild it if you'd like to send me a
new version
of the patch. Thanks - John
ARGH Not so good. Asterisk now segfaults on start up :((( - John
Now that is a behavior I'm not seeing,
This has been fixed in the 1.6.1 SVN, and you will have to back
port a patch until these changes are rolled into another release. I was
disappointed that more bug fixes were not included in 1.6.1.1.
-Jonathan
Asterisk 1.6.1.1 was released
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Ernest Byaruhanga
Sent: Thursday, July 02, 2009 4:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Nortel pbx dtmf issues
folk,
I have a basic config for AEL syntax highlighting for Kate if you would
like it.
- Brad
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, July 10, 2009 8:46 AM
This is actually working as designed. You need to use type=peer in order for
call-limit to work properly, which in turn is what allows hints to work
properly.
- Brad
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gagnon
Sent: Tuesday, August
If you already have the IP in a file, why don't you set it up so the
file itself says: externip=xx.xx.xx.xx and then do a #include in
sip.conf for the /etc/myip file? I believe you'll have to do a sip
reload either way (which can obviously be part of your cron job) if
you're not already, but
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, August 23, 2006 8:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hint extension issue - bug?
On Wednesday 23 August 2006 06:56, Watkins, Bradley wrote:
This is actually
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, August 23, 2006 9:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hint extension issue - bug?
On Wednesday 23 August 2006 08:48, Watkins, Bradley wrote:
It's not a bug. When you use type
I may have to eat my words, then. This is the case with trunk, and I can't
recall the last time I built a 1.2.x system. I could have sworn that behavior
didn't change, but I've been wrong before.
- Brad
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
In the forthcoming 1.4, you can tell the Queue application to run an AGI just
before sending the call to the destination. In the AGI, you can use the (also
new in 1.4) MEMBERINTERFACE channel variable to determine the destination.
Of course, that's not a solution now since 1.4 is not even
Did you ever try to get it working on any 1.6.x releases? I hacked at
it a bit and it didn't seem to be working, though I could have been
doing something wrong. I was, after all, reading the manual... ;)
I'm glad to hear someone successfully doing it, as it's something I've
wanted to play with
You will need to change the type=friend to type=peer and
also define call-limit to some value (it can be large if you don't care about
the actual limit). That should fix hints for you.
Regards,
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall,
Eric
September 2006 12:31, Watkins, Bradley wrote:
You will need to change the type=friend to type=peer and
also define
call-limit to some value (it can be large if you don't care
about the
actual limit). That should fix hints for you.
But if you have it set to 1 then busy status won't
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Crocker
Sent: Thursday, September 28, 2006 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk - Tekelec T6000
(Vocaldata, voiss)
You didn't say, but my guess is you are using either a 4-port or 2-port
Digium card, right?
What do the contents of /etc/modprobe.d/zaptel look like?
You will probably find that there isn't an entry like:
install wct4xxp /sbin/modprobe --ignore-install wct4xxp $CMDLINE_OPTS
/sbin/ztcfg
I put
Does that entry exist also in e164.arpa (the
default)? Have you tried explicitly pointing it at e164.org
instead?
FWIW, I see nothing in particular wrong about your usage,
but make sure we're talking about the right trees here.
Regards,
- Brad
From: [EMAIL PROTECTED]
I playing a bit with this, it seems that if you use the new syntax it
works:
exten = _[a-z].,3,VoiceMail(${EXTEN}|u)
You can, of course, also use the b, j, s, and g flags.
Even using the VoiceMail(u${EXTEN}) still elides the 'j'.
Regards,
- Brad
-Original Message-
From: [EMAIL
The only phone that I know of that has a 10/100/1000 switch in it is the
Cisco 7971G-GE:
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0900a
ecd801c5c4a.html
It is one heck of a phone, but the price is relatively astronomical (list
price is roughly $1100 each).
- Brad
Ooops... I should also mention that apparently they don't support SIP (I
was just looking). I saw them demoed by Cisco on a CallManager box awhile
ago. I guess I just assumed that a newer phone like that would have SIP
firmware available as well.
Sorry for any confusion.
- Brad
-Original
What revision of card is the new one? It sounds like you have one of the
new Rev I cards and you aren't running either 1.0.9 or CVS HEAD. Either of
these will solve your problem if I am correct.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Title: Message
I
believe you have to upgrade to 5.3 in order to go from unsigned to signed
executables. Once you're at 5.3, you can go directly to 7.5. I did
this recently with a couple of 7960s I had in the lab and it worked
perfectly.
Regards,
-
Brad
-Original
That depends on what you mean by default. The supplied sample
extensions.conf contains the priorityjumping=no by default, but if this
parameter is absent then the default is to jump n+101.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I have several servers using them, but I only needed to download them
directly from you just once. I replicated the bits myself. The magic of
these advanced technologies... ;)
I could go download them a few more times if it would make you feel better.
All kidding aside, I don't think I ever
Could you perhaps post your dundi.conf for both boxes? I'm afraid this
message doesn't mean anything to me, but I have about a dozen boxes doing
DUNDi peering so I know what the config should look like. But it's
basically always worked for me.
Regards,
- Brad
-Original Message-
From:
At the moment I'm out of the office, but when I return I'll be certain to do
that. Note that my solution is different from what you are working on with
regexten, though I suspect some of the challenges that I've faced and
overcome are not. I'm actually using UltraMonkey for load-balancing and
problem when
loadbalancing with Ultramonky? As I understand it LVS does not properly
support SIP in that it doesn't always use the same source path.
regards,
David
On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
At the moment I'm out of the office, but when I return I'll be certain
to do
.
Regards,
- Brad
_
From: [EMAIL PROTECTED] on behalf of David Thomas
Sent: Fri 3/17/2006 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: DUNDi Halfway and CLUSTERING
On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
Do
).
Regards,
- Brad
_
From: [EMAIL PROTECTED] on behalf of David Thomas
Sent: Fri 3/17/2006 5:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: DUNDi Halfway and CLUSTERING
On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
I
No flames here as I realize that there are plenty of limitations with MySQL,
but if you're using the current GA of it views is not one of them.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: Sunday, March 19, 2006 4:15
What version of Asterisk are you running?
The reason I ask is that I think I remember a fix for this on the
svn-commits list awhile back.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Tuesday, March 21, 2006 4:52 PM
I apologize, but the fix I was thinking of wasn't directly related to this.
It was in app_voicemail.c, but related to using the channel's context for
the Directory application. The fix for your issue may be indirectly
related, though. I would open a bug.
Regards,
- Brad
-Original
Did you download it from asterisk.org? I didn't have the latest -addons,
but I just downloaded it and it does have Copying which contains the GPLv2.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. J.
Meidlinger
Sent: Wednesday, March
I don't think there's any kind of (significantly small, anyway) limit. I
have over 300 users at one site in voicemail.conf and no issues there.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil
Sent: Wednesday, March 22, 2006
but
when I call the users above 70 it prompts me User Not Found.
Any idea regarding this?
Thanks,
Ryan
At 04:12 AM 3/24/2006, Watkins, Bradley wrote:
I don't think there's any kind of (significantly small, anyway) limit.
I have over 300 users at one site in voicemail.conf and no issues
That implies that the 2850 has a standard molex connector anywhere inside of
it, which is not the case.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Davies
Sent: Wednesday, March 29, 2006 6:34 AM
To: [EMAIL PROTECTED]; Asterisk
The OP was referring to how sox interprets filename extensions. In that
case, Kevin's .raw and .sw extensions are correct.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, April 19, 2006 12:56 PM
To:
Not to mention the obvious, and this may not help your situation, but if
you were (or are) using templates it would be a one-line change.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Piper
Sent: Tuesday, May 02, 2006 12:49 PM
The answer is to create a separate context for your regextens (or, more
appropriately, name it in sip.conf and let chan_sip create it) and then
include that context in your dundi_local context where you have the
dialing information.
Regards,
- Brad
-Original Message-
From: [EMAIL
I'm not sure if you have considered this, but if you were using SIP
between the Asterisk servers you can definitely achieve this using
X-headers.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, May 11,
I have not found a way to do this via the Polycom
configs. However, what I do is just ensure that the callerid of an inbound
call is set so that the recorded number on the Polycoms is a valid callback
number (i.e., prepend '9' or '91' depending on the inbound
CallerID).
Regards,
- Brad
How about the dialplan function SIP_HEADER?
-= Info about function 'SIP_HEADER' =-
[Syntax]
SIP_HEADER(name)
[Synopsis]
Gets or sets the specified SIP header
[Description]
Not available
dtw-test-asterisk-001*CLI
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
Yes, the (newly rewritten, but compatible with older AEL files) version
of AEL that is currently in trunk is staying and I think personally it's
a big step forward. I still haven't gotten used to it myself, but then
the folks who used to write everything in assembler probably took a
while to get
I've never attempted to use this feature, so I can neither confirm nor
deny whether it works/doesn't work/used to work/etc.
But what I find really odd, is that the code doesn't even appear to try
and parse astdb when it's loading the config, at least insofar as I
can tell. A quick grep -i astdb
making it into chan_sip.c
However, the option *did* make it's way into sip.conf, so I guess that
the real bug is that the option is in sip.conf.
Bummer.
Devels: Any chance of getting 3359 re-opened and put into asterisk ?
Julian
Watkins, Bradley wrote:
I've never attempted to use this feature, so
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