We tried with MS Exchange but couldn't get it to work (MS Exchange
doesn't support a master account).
CP
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Thursday, October 18, 2007 11:20 PM
To: Asterisk Users Mailing List -
Hi Doug,
What combination of bootrom, sip version and FTP server are you using?
There is a combination with vsFTPd which can cause this sort of problem.
CP
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Sent: Thursday, September 27, 2007 3:30 PM
Hi Matt,
We have one and it works very well - usual Polycom quality, as others
have attested. The only thing we have noticed is a reluctance to
download its config files via FTP when using a VLAN tag.
CP
Matt wrote:
Hi,
Has anyone here ever used a Polycom IP 4000 Soundstation SIP
Conference
Hi David,
Disable URL dialing (url-dialing in the feature/ section of sip.cfg.
CP
Klaverstyn, David C wrote:
Hi All,
I have a site using Polycom 501 phones and for some reason the caller
ID of the phone number is coming up as sip:number@ip of server
Does anyone know why? It seems
And yet, it's shorter than your HTML/image-ridden sig. :-)
CP
Wiley Siler wrote:
Disclaimer at the bottom still looks ridiculous even in Spanish… LOL
*Wiley E. Siler
**Director of Information Technology*
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
I use Bluefish, and have developed a syntax-highlighting template for
Asterisk conf files, if you're interested.
CP
Steve Finkelstein wrote:
This might be of some assistance:
http://www.voip-info.org/wiki/view/vim+syntax+highlighting
- sf
Olivier wrote:
Hi,
New to Kubuntu and Linux,
Try the 'g' option to VoiceMail().
CP
Stephen Bosch wrote:
Hi:
I have a user saying that the volume of voice mails is too low.
Is there a way to tweak the recording level for voice mail?
-Stephen-
___
--Bandwidth and Colocation provided by
That's the way we want to go, but have been unable to divine the correct
settings for getting it working with MS Exchange.
CP
Tim Panton wrote:
If I were starting a project now, I'd
take a look at the (newish) support for IMAP storage for voicemail.
mount -o intr,nolock ought to do the trick. we're using those
options now, but thankfully haven't had reason to find out if they work
or not yet.
CP
Doug Garstang wrote:
No, you can get Asterisk and NFS to work fine together. It was in my
past job, so I can't remember the exact
It will stall asterisk - ask me how I know.. :-)
CP
Gordon Henderson wrote:
On Tue, 24 Apr 2007, Forrest Beck wrote:
I've heard there are problems using NFS as a storage device.???
I've used NFS for many many years on 100s, maybe 1000s of servers in this
time. It's great. Just
Why not export an NFS mount from one server to the other? That's what we
do.
CP
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest
Beck
Sent: Tuesday, April 24, 2007 5:28 PM
To: Asterisk Users List
Subject: [asterisk-users] Voicemail on
Hi John,
Try 1.4.2 - there was a bug in earlier versions that produced the
symptoms you describe (http://bugs.digium.com/view.php?id=8848, and
various related ones).
A.
John Hughes wrote:
Playing with hints/presence/BLF on asterisk I've made the following
discoveries.
1. The wiki at
to the other then reload asterisk nightly. The biggest Con to
this is I have to be sure my dialplans don't get different. The
user's voicemail wouldn't be available until their primary server is
back up, but that's OK.
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web
any specifics.
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE
Greetings,
Very occasionally, we have a complaint from a user that a portion of
a voicemail message is very speeded up - like when you press the fast-
forward button on an old-fashioned tape dictaphone. This affects both
the server-stored and emailed copies of the message. I have a sample
We have it working fine on an SPA-3000.
CP
On Feb 5, 2007, at 10:42 PM, Joseph wrote:
I've a problem with inserting a pause and dialing additional numbers
when going through Sipura-3000
exten = _12,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww18))
D() doesn't work as it sends the DTMF tones
Hi there,
We traced this issue to transfers (see http://bugs.digium.com/
view.php?id=8848). On Monday, I will be attaching the various debugs
to the case as requested.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http
of this
opportunity to obtain Asterisk bootcamp training in the Pacific
Northwest.
Space on the course can be booked via the Digium web site at
http://www.digium.com/en/training/locator/enroll/46.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http
Here's how to unsubscribe:
First, ask your Internet Provider to mail you an Unsubscribing Kit.
Then follow these directions.
The kit will most likely be the standard no-fault type. Depending on
requirements, System A and/or System B can be used. When operating
System A, depress lever and a
Is anyone else having trouble getting a Polycom IP4000 (running SIP
1.6.7 and BootROM 3.1.3) to download its configuration files from a
vsftpd 2.0.1 server? We have 100+ IP501s that manage this without
problems, but the IP4000 keeps timing out.
We have opened a case with Polycom, but they are
Just 'sox -v 1.5 beep.gsm loudbeep.gsm' ?
CP
On 2-Dec-06, at 11:29 AM, Peder @ NetworkOblivion wrote:
We've had a few people complain that the beep before leaving a
voicemail is not loud enough and too short. Does anybody have a
recorded beep that they can share, that is a little louder and
IIRC, menuselect requires ncurses-devel (or your distro's equivalent).
CP
On Dec 1, 2006, at 7:05 AM, Doug Crompton wrote:
No, no menuslect on system beside *
I unzipped it, ran configure, then make (or make menuselect) they both
give the same immediate error 3.
From what I see with 1.4.x
Please don't cross post FS items to *-users - that's what *-biz is for.
CP
On Nov 24, 2006, at 10:45 AM, Mark Phillips wrote:
Hi all,
I have a surplus Sangoma 10 port FXO card for sale. This model could be
upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by
changing the
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
CP
On Nov 22, 2006, at 8:40 PM, ram wrote:
Hi
where can i buy that Book
Ram
On 11/22/06, Patrick [EMAIL PROTECTED] wrote: On Wed,
2006-11-22 at 15:45 +0530, ram wrote:
[snip]
Nov 22 15:43:23 WARNING[14623]:
Just use Set(CALLERID(name)) in your dialplan - that's what we do.
CP
On Nov 23, 2006, at 12:00 AM, Eric Bishop wrote:
When we have calls that originate click-to-daial apps that use the
manager interface they always originate from asterisk is there any
way to change the from name?
you file a bug report you might want to check
to see if there are changes to the way hints are implemented in 1.4.
It might be a configuration problem rather than a bug but I have not
had
time to look into it.
John
On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote:
Hi
http://bugs.digium.com/view.php?id=8405
On Nov 22, 2006, at 9:11 AM, Anthony Rodgers wrote:
Thanks, John - this confirms what we are seeing. 'show hints' output
isn't changing, so it looks like a bug. I'll open one and see what
happens.
A.
On Nov 21, 2006, at 5:44 PM, John Lange wrote
We narrowed this down to when the 'New Call' softkey was used to
initiate the call. When this key was used, the corresponding 'Placed
Calls' entry wouldn't work. Any other method of placing the call does
work.
An upgrade to 1.6.7 fixes the issue.
CP
On Nov 16, 2006, at 4:34 AM, John Marvin
Thanks, Noah - we'll try 1.6.7 and see if the problem goes away.
CP
On 15-Nov-06, at 11:55 AM, Noah Miller wrote:
Has anyone noticed that attempting to place a call from the
Placed
Calls list on a Polycom IP501 by pressing the 'Dial' softkey
sometimes
simply returns the phone to
Hi James,
We're running SIP 1.6.6.0036 on the 3.1.3.0131 BootROM.
Did you come up with any reason/fix for this?
CP
On Nov 13, 2006, at 11:00 PM, James Andrewartha wrote:
Anthony Rodgers wrote:
Greetings,
Has anyone noticed that attempting to place a call from the Placed
Calls list
Have you tried setting up a hint for a ZAP channel?
exten = foo,hint,ZAP/bar
Then make a directory entry for foo in your Polycom directory for foo -
just as you would if the hint was for a SIP channel.
CP
On Nov 14, 2006, at 4:26 AM, Robert Jenkins wrote:
Hi,
I've recently got some
Greetings,
Has anyone noticed that attempting to place a call from the Placed
Calls list on a Polycom IP501 by pressing the 'Dial' softkey sometimes
simply returns the phone to the idle screen? It is not related to the
number being dialed, as we have observed two entries for the same
number,
Here's how to unsubscribe:
First, ask your Internet Provider to mail you an Unsubscribing Kit.
Then follow these directions.
The kit will most likely be the standard no-fault type. Depending on
requirements, System A and/or System B can be used. When operating
System A, depress lever and a
the
Asterisk server like a CO; calls to Asterisk from the Nortel are on
an NI2 tie-trunk to allow the Nortel to send CallerID to the Asterisk
server.
Hope this helps - I have the Nortel config we used in a PDF if you
need it.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North
Can you be more specific? What sort of linkages are available between
the two offices?
CP
On 22-Oct-06, at 10:38 PM, dthurn wrote:
What's the best way to connect an Asterisk PBX to a Nortel MICS PBX.
I have two offices that I want to link together.
TTFN
Greetings,
This is my annual post-Astricon attempt to start an Asterisk User
Group in the Vancouver, BC, area. If you are interested, please reply
off-list.
Regards,
--
Anthony Rodgers (CunningPike)
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http
resellers, carriers and call-centers who are using Asterisk to
support their non-telecom-related business - I don't envisage any
geographical limitation to the group (there seem to be few enough of
us as it is!).
If you are interested, please let me know off-list.
Regards,
--
Anthony Rodgers
IMHO, FTP really is the way to go - you get the ability to have the
phones detect config file changes and automatically reboot, and you
get the ability to upload logs, custom configs and directories from
the phones.
We use vsftpd, with the default user and password for the phone.
CP
On
We had/have this problem, too - we eventually got it working (just by
constantly rebooting it), but it seems that something's not working
properly somewhere..
Can you look in your phone's boot log and see if you are getting any
errors? We were seeing errors relating to the phone not
Hi Eddie,
Connect to irc.freenode.net, and then type this:
/msg nickserv register password
nickserv will tell you that your nick is now registered.
Then type this:
/j #asterisk
Say hi to CunningPike when you get there.
CP
On 24-Oct-06, at 1:12 PM, Eddie Johnson Jr wrote:
Hello Dovid,
Did you compile and install these in the correct order:
zaptel
libpri
asterisk
CP
On 23-Oct-06, at 5:47 AM, ram wrote:
Hi all
I have installed 1.2.12.1 in FC5 with libpri.1.2.4
when i start
iam getting the following error and it quits
== Registered channel type 'Local' (Local Proxy
Like the one that comes with it?
[EMAIL PROTECTED] ~]$ sudo more /etc/init.d/op_panel
#!/bin/bash
#
# chkconfig: 2345 99 15
# description: Flash Operator Panel
# processname: op_server.pl
# source function library
. /etc/rc.d/init.d/functions
DAEMON=/usr/local/op_panel/op_server.pl
OPTIONS=-d
Local government office with approximately 100 sets (going to 600):
593 extensions (1241 priorities) in 88 contexts
CP
On 10-Oct-06, at 1:16 PM, Steve Murphy wrote:
Hello!
In my relentless quest for knowledge, I pose this question: who's got
the biggest
dialplans, and how big are these
Hi Eric,
Here's all we had to do:
1. Make sure that the 'Presence' feature is enabled in your phones:
feature feature.1.name=presence feature.1.enabled=1.. in
sip.cfg (or maybe ipmid.cfg, depending on the age of your SIP
application)
2. Create a hint priority in extensions.conf
Likewise, Ronnie, we have 2 PRIs going to an 11C - let me know if I can
help.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Sep 27, 2006, at 2:42 PM, Savoy, Kevin - Williston, ND wrote:
Ronnie
Hi Mike,
It's done using the digitmap feature of sip.cfg - email me offlist or
come on #asterisk and I can help you with the specifics.
CP
On 18-Sep-06, at 11:08 AM, Mike wrote:
I'm trying to make the Polycom 501 go off-hook (in speaker phone
mode) when any digits is dialed and the
I concur:
HUDLite - couple of days, unanswered forum postings, never got it
working
FOP - few minutes, worked right away
YMMV,
CP
On Sep 14, 2006, at 7:45 AM, Brodie Macleod wrote:
Yeah there are some problems with the docs, and the product itself
isn't very
impressive -- still bugs that
This looks like a networking issue - asterisk isn't receiving any
replies to signaling packets and assumes that the UA is no longer
reachable.
CP
On 8-Sep-06, at 10:33 AM, Noc Phibee wrote:
anyone know this error ??
Noc Phibee a écrit :
Hi
today, i have a big problems with my
Or better yet, set dialplan.impossibleMatchHandling to 2. This should
disable earlydial altogether.
CP
On Sep 8, 2006, at 2:49 PM, Eric ManxPower Wieling wrote:
Mike wrote:
It's not a silly idea, I've been doing some sniffing and debugging
with my
limited knowledge of the whole process.
With respect, the problem is with your numbering plan..
CP
On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote:
I found a problem in blind transfer:
I have an extension number 601 and I have an extension 6014
If I get a call on 615 (snom) and transfer to 6014 it works, since
snom
Here's what we do:
[agent-login]
exten = s,1,NoOp(${AgentUser})
exten =
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})
exten = s,3,Wait(1)
exten = s,4,Playback(agent-loginok)
exten = s,5,Hangup
exten = s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
exten =
Westany speaks biz
CP
On Aug 30, 2006, at 9:50 AM, Stuart wrote:
Westany, the Asterisk voice experts, announce their first Russian
voice for
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
There is a good page on the wiki about this:
http://www.voip-info.org/wiki-Asterisk+non-root
CP
On Aug 14, 2006, at 6:44 PM, Forrest Beck wrote:
Does anyone have a listing on file/directories that asterisk needs
ownership of to run as a user other than root?
I know about the major items ---
Empty line keys will be filled with speed dial entries in the phone's
directory - when creating a directory entry, set the speed dial value
to 1 for the first, 2 for the next.. etc.
CP
On 16-Aug-06, at 11:23 AM, Warren ((mailing lists)) wrote:
I just got my first IP601 and put
Hi Angel,
We have two DL360s with a TE410P in each one - we had to disable USB to
get the PCI slot to have an IRQ to itself.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Aug 2, 2006, at 6:38
Yup - burned us a few times, too - on IP501s as well.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Aug 3, 2006, at 6:42 AM, Bill Gibbs wrote:
I thought I was the only one!!! I actually
This looks like a dialplan problem - do you have a match for
0109687348 in the zap-in context in your dialplan?
A.
On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote:
Dear All,
I have a strange problem in recieving calls on the pri the zaptel
is green and everything seems very well,
Hi Kai,
This is what we do:
[agent-login]
exten = s,1,NoOp(${AgentUser})
exten =
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})
exten = s,3,Wait(1)
exten = s,4,Playback(agent-loginok)
exten = s,5,Hangup
exten = s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
The 'o' option to the Dial() command, along with using blind transfers,
fixed this problem for us.
A.
On Jul 25, 2006, at 11:25 AM, Douglas Garstang wrote:
I have three phones here with extensions 2944093, 3254103 and 9220371.
2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We
Aha - get rid of the leading comma for each entry..
= ,Front Desk
= ..
A.
On Jul 13, 2006, at 1:00 PM, Kevin Savoy wrote:
I've X'd out the extensions and passwords but this is all I have in
there.
Thanks
[default]
=,,Front Desk,,
Try having nothing after the name in your voicemail.conf:
1234 = 1234,The Marquis de Sade
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jul 12, 2006, at 11:17 AM, Kevin Savoy wrote:
I have
then get:
[EMAIL PROTECTED]
Which of course fails because that address doesn't exist.
Any other ideas?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Thursday, July 13, 2006 2:24 PM
To: Asterisk Users Mailing List - Non-Commercial
We have just come through our busy tax season for our tax line queue on
1.2.1 with zero problems :-)
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jul 13, 2006, at 12:41 PM, Rich Adamson wrote
Yes - and it seems to prevent presence hints from working until the
phone is rebooted..
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jun 26, 2006, at 9:28 AM, Douglas Garstang wrote
Actually, if his MTA is configured properly, it shouldn't happen at
all.
A.
On Jun 22, 2006, at 9:32 AM, Doug Geary wrote:
Should only happen once if his email system is config'd in a standard
method. Otherwise just *plonk* his address.
-Original Message-
From: [EMAIL
Care to share your Nagios plugin?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jun 22, 2006, at 9:53 AM, Curt Shaffer wrote:
Does anyone out there have a recommendation for tools
of email from outside your organization from
people who expect a response, it is helpful to us (and them) if they
receive OOO notifications.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jun 22
of 80ms or 80% packet loss for a
warning
and 100ms or 100% packet loss for critical. The perfdata is then
passed to
perfparse for graphing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Thursday, June 22, 2006 2:02 PM
To: Asterisk
Hi Philippe,
Blackberries can't play sound file attachments - wish they could.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jun 1, 2006, at 2:33 PM, Philippe Lindheimer wrote:
Aaron
Is there any chance you're connecting to a remote share using CIFS?
What does slabtop look like on your machines?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 29-May-06, at 8:35 AM, Attilla
Hi Jamie,
Take a look at the dialstring in your sip.cfg - you'll need to adjust
this to match your local dialing plan.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 26-May-06, at 2:49 AM
That would be we 48, no? :-)
I think this thread needs an AK-47 now...
A.
On 24-May-06, at 12:33 PM, Paul wrote:
If I had 47 siblings it could also mean us 48
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
Greetings,
I am trying to gauge the level of interest in an Asterisk users'
group in Vancouver, BC (or in BC in general). If you would be
interested, please reply off-list.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed
Or use the newer syntax for Voicemail:
exten = s,n,Voicemail([EMAIL PROTECTED]|su)
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 7-May-06, at 11:07 AM, Ira wrote:
At 04:33 PM 5/6/2006, you
Done with timeout=600 and queue-thankyou=path/to/sound/file in
queues.conf
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 8-May-06, at 10:27 AM, Matt wrote:
Hi,
I know that I can have
) - the rest of you will have to wait until
they're finished :-) but I do intend to release a bunch of monkey-level
helpdesk scripts that I am working on in the near future for managing
basic MAC requests.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http
Looks like a timing problem - zaptel.conf and zapata.conf, please.
A.
On Apr 25, 2006, at 3:05 AM, Nico Giefing wrote:
Hello,
I get an Error every minute on the second card of two installed TE410P
Cards in our System.
The error is:
PRI got event. HDLC Abort (6) on Primary D-channel
,SetCallerID(${CALLERIDNAME} 604998${CALLERIDNUM})
exten = s,6,Goto(20)
exten = s,10,SetCallerID(${CALLERIDNAME} 604990${CALLERIDNUM})
exten = s,11,Goto(20)
Hope this helps - let me know if you need more details.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http
this helps.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Apr 20, 2006, at 6:37 AM, Bart van Daal wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hi Domenico,
We're using RHEL 4 ES with no obvious issues
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Apr 21, 2006, at 3:59 AM, Mimmus wrote:
Hi,
I'm planning to install a new Asterisk
this issue? What does your slabtop look
like?
Any thoughts or ideas would be appreciated.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
___
--Bandwidth
that
allows uploads is available to store a back-up copy of the directory
or its contents will be lost when the phone reboots or loses power.
Hope this helps.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
Does anyone enjoy these?
It's funny - I see people being flamed for asking Asterisk questions,
but not a murmur about this stuff...
On Apr 13, 2006, at 5:26 PM, Bob's Leaky News Service wrote:
Will VoIP be Next?
snip verbal diarrhoea
___
for this? :-)
Then, in extensions.conf, set a hint for the _watched_ extension like
this:
exten = 2348,hint,SIP/2348
Let me know if you have any more questions.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 3-Apr-06
Interesting - we didn't find this on either the 501s or the 601s
A.
On 3-Apr-06, at 1:11 PM, Darrick Hartman wrote:
Additionally, (at least on the Polycom 600's)
you need to reboot your phone for this to take effect.
Darrick
--
Darrick Hartman
DJH Solutions, LLC
I tried to get a government/enterprise SIG or UG off the ground a
number of months ago, with very limited success. If there is sufficient
interest now, I could be persuaded to try again.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http
--- Downloadable D-Channel Handler C
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Mar 22, 2006, at 12:48 PM, Steve Rawlings wrote:
I've followed the post below and have just acquired a second-user
Option
Hi Greg,
I'll dig it out - we only expand the outgoing callerID to 10 digits for
external (PSTN) calls, so we don't have the CID issues you mention.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
and is used only for calls from Asterisk to the
Nortel.
If you need more specific details, let me know.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Mar 22, 2006, at 3:21 PM, Greg Camp wrote:
Hello
Many thanks, Russ - I'll give this a try.
Thank goodness a) for test servers and b) for the ability of Linux to
rollback with a simple change to grub.conf :-)
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org
AFIAK, they can't - we would like to do the same thing, but it's not
possible with patching the source.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 10-Mar-06, at 7:56 PM, btb wrote:
can
=international
trunk=yes
The iax.conf from the obelix server would be similar. Hope this gives
the idea OK - let me know if you need any more information.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
-itsresourcec,2006-03-03 13:13:36,2006-03-03
13:13:43,2006-03-03 13:16:02,146,139,ANSWERED,DOCUMENTATION
4 UAs are dialed - only one answered the call - only one CDR record.
Hope this helps.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
-22.0.2.EL, zaptel compiles
just fine.
This behavior is true for both zaptel-1.2.1 (shown above) and
zaptel-1.2.4.
Thoughts?
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
Hi Bradley,
Yes, I experienced quite a lot of echo with my IAXy, until I switched
analog handsets - in my case, it was severe acoustic coupling in a
cheap handset.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http
Hi Yusuf,
All our * boxes have SELinux installed and active - we haven't had to
make any changes to the default SELinux config to make * work properly.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
Are you sure you're supposed to be using EM?
On Feb 24, 2006, at 5:39 AM, Nitin Joshi wrote:
Hi All,
I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its
connected directly to the PSTN. But I am unable to make outbound calls
on the zap channels. The light on the card is
You'll likely find Asterisk itself even more of a challenge then.
On Feb 15, 2006, at 1:29 PM, roswel ajf wrote:
hi,
To post, and to reply to a post, i have to goto my email. Now, if
there is a
web interface to these mailing list, things would be easier.
Your output looks like you have 3 cards, two of which are sharing
interrupts - or am I missing something?
On Feb 10, 2006, at 7:04 AM, Gerard Saraber wrote:
So nobody heard these before? or did I do something stupid that anyone
should know and nobody wanted to yell at me for it ;)
On Wed,
On Feb 8, 2006, at 9:27 AM, Greg Camp wrote:
Now, our latest two issues:
1) When a user on the Nortel makes a call to a user on * a 10-digit
callerid value shows up on the SIP phone instead of the users
extension.
Has anyone encountered this and found a work-around? It's been
suggested
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