On Fri, Sep 3, 2010 at 6:11 AM, mattias m...@mjw.se wrote:
Can i run asterisk on a openvz vps or do i need a kernel?
I dont use dadi
--
Works just fine for our voicemail server (~450 users).
CP.
--
_
-- Bandwidth and
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote:
Thank you Andrew,
I will check it out. We are currently running 1.4.
-Matt
On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote:
Remote Party ID in trunk, it works There are hacks for other
On Thu, Jun 3, 2010 at 6:16 AM, Gilles codecompl...@free.fr wrote:
Hello
I just read this article and would like some feedback from
experienced Asterisk users:
===
Failed open source VoIP deployment leads to hosted VoIP strategy By
Jessica Scarpati
snip
https://issues.asterisk.org/view.php?id=6643
CP
On Thu, Apr 1, 2010 at 7:36 AM, Ondrej Valousek webs...@s3group.cz wrote:
Hello,
Did anyone manage to force asterisk to put Remote-party-ID attribute on
the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
B displayed on
On Fri, Jan 22, 2010 at 7:50 AM, Mike l...@virtutel.ca wrote:
I know having Asterisk aware of Polycom Do No Disturb state wasn't working
before (1.4), but is this working in any recent version? Is there any
custom way of doing this?
Our Asterisk servers (1.2 and 1.4) get SIP response 603
On Tue, Dec 8, 2009 at 7:37 AM, Noah Miller noahisaacmil...@gmail.com wrote:
Hi -
I am having echo issues on our Asterisk box using a PRI circuit. I was
using the software echo cancellation and that helped a bit but didn't solve
it completely. So I went and bought a Digium echo cancellation
On Mon, Oct 5, 2009 at 11:34 AM, Ken D'Ambrosio k...@jots.org wrote:
Hey, all. Just wondering if there's a GUI out there -- preferably OSS,
but I'll take what-have-you -- that
a) can run on an Ubuntu/Debian box, and
b) allows a receptionist to see what calls are in-process, and forward
This is so wrong it's not funny. The caching of DNS SRV records acts in
its favor when it comes to failover - the UAs already have the
information they need in their resolver cache to perform the failover
without having to make another DNS query.
The TTL you need to worry about is that of the SIP
Just in case anyone is having DNS SRV timeouts with their Polycom
phones, the following Polycom KB article should help:
http://knowledgebase.polycom.com/kb/search.do?cmd=displayKCdocType=kcexternalId=12856sliceId=SAL_PUBLIC_1_2dialogID=7620671stateId=1
We have set
Oliver,
We use DNS SRV records combined with short TTLs to provide failover.
Thankfully, we have only used it when moving phones from one server to
another in preparation for upgrades, but it worked like a champ then.
CP
On Wed, 2008-09-10 at 15:02 +0200, Olivier wrote:
Hi,
I'm planning to
Well, that was sorta my point.
CP
Steve Edwards wrote:
A quick grep through the Asterisk (1.2.28) sources shows res_monitor using
soxmix if the channel variable MONITOR_EXEC is not defined -- but nothing
in app_voicemail. Am I missing something?
Hi Daniel,
I'm intrigued by this and wanted to try it out - but I'm wondering how
you get Asterisk to call sox at all during Voicemail()? Our server
doesn't even have sox installed, so I'm not sure how to go about
tricking Asterisk into running a different one.
CP
Daniel Hazelbaker wrote:
What type of Nortel? How are you connected to the Nortel?
CP
Eugen Soare wrote:
Well I am entering into a realm that I don't know.
3 sites with Asterisk
1 site with Nortel
Asterisk/Sip calls working fine between the 3 sites.
Asterisk to Nortel set calls working fine. (call comes
Have you set the VLAN tag on the phone?
CP
Lee, John (Sydney) wrote:
Hi all,
I have been googling and testing without any luck and would appreciate
any guidance from anyone.
A port has already been configured on the CISCO switch with the
following:
interface FastEthernet2/0/1
Try 'ip4000_1' instead of '207' for your address.
CP
Kevin DeGraaf wrote:
I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on
a flat local network.
I followed the provisioning guides that I found on the Web, and I have
the phone downloading bootrom.ld, sip.ld, and a
Sounds like good security practice to me. YMMV.
CP
sandeep.s wrote:
Hi,
my sip phone is unreachable for external network(global ip)
Thanks,
sandeep.s
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
Hi Michelle,
We added to the bounty for this feature some time ago[1], and had a
developer all lined up. He was unwilling to proceed, because Digium said
that our work would never get accepted because they were already
working on it. The IMAP support in 1.4 must have been it - doesn't work
for
Well, there you go then - either add /usr/sbin to your path, or provide
a full path thusly:
/usr/sbin/asterisk -r
CP
Robert McNaught wrote:
not in path
[EMAIL PROTECTED] echo $PATH
/usr/kerberos/bin:/usr/lib/courier-imap/bin:/usr/local/bin:/bin:/usr/bin:/usr/X11R6/bin:/home/admin/bin
Is /sbin in your path?
CP
Robert McNaught wrote:
my problem is that a non-privileged user, eg admin, cannot log in and
connect to the console by issuing the following
[EMAIL PROTECTED] asterisk -r
bash: asterisk: command not found
[EMAIL PROTECTED] whereis asterisk
asterisk:
Try dtmfmode=inband
CP
Lyle Giese wrote:
I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13
with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of
Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog
phones via Adtran chan bank. When I went to
Disable URI dialing on your phones.
CP
Rob Schall wrote:
I have an asterisk 1.4 setup with a PRI installed and working. We are
using a Polycom 501 to test the setup..
Inbound calls work great as do phone to phone calls.
However in all cases, the caller id is a bit odd. It shows:
21 matches
Mail list logo
