can do
something like this:
[phones-in-account1]
include => downstream-phones
exten => 911,s,Goto(DialViaAccount1)
[phones-in-account2]
include => downstream-phones
exten => 911,s,Goto(DialViaAccount2)
etc.
--
Dave Miller http://www.justdave.net/
Sys
e I had problems with their Chicago node dropping
out. But I get pretty good connections with minimal latency from New
York, despite sending the packets right past the Chicago one. :)
Strange, but it works for me.
--
Dave Miller http://www.justdave.net/
System Adminis
the problem.
You should be able to tell it to log to a file in addition to the
console in logger.conf. Something like:
full => notice,warning,error,verbose
Then it should show up in /var/log/asterisk/full and you wouldn't need
to keep a session open to the console to see it
my
first guess.
--
Dave Miller http://www.justdave.net/
System Administrator, Mozilla Corporation http://www.mozilla.com/
Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/
___
--Bandwidth and Coloc
uot;8" line. According to the
> 2.1.0 admin guide that means the second week of the month but none of
> the guides before that mention this as a valid option.
Thanks! One question I have... with this applied (and even with the
original config I had before changing it to this), the
used to use
before we had Asterisk. :)
The story is likely what hardware you have it running on. If you expect
your phone system to be an enterprise-class PBX, it needs to run on
enterprise-class hardware, not some leftover 486 box from the back closet.
--
Dav
m and would not address the problem
> despite repeated calls.
Also of note is that the time zone can also be set via DHCP, and if it
is, that can't be overridden in the phone, either.
--
Dave Miller http://www.justdave.net/
System Administrator, Mozilla Corpo
g an NTP server, so
you need to set one.
--
Dave Miller http://www.justdave.net/
System Administrator, Mozilla Corporation http://www.mozilla.com/
Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/
e was a security update for Asterisk released yesterday which
addresses a denial-of-service class vulnerability in which malformed SIP
packets could cause asterisk to crash. Our server also randomly died
once on Monday morning with no apparent cause. It's possible someone
was exploiting t
! Hanging up.
I've temporarily worked around it by switching our inbound provider to
use SIP instead of IAX, but that's not an ideal solution.
--
Dave Miller http://www.justdave.net/
System Administrator, Mozilla Corporation http://www.mozilla.co
Dave Miller wrote on 4/26/07 11:46 AM:
> We upgraded our asterisk server to 1.2.18 last night to pick up the
> security update. Since then, any calls coming in on IAX2 links get
> dropped if they try to enter a MeetMe conference room.
>
> The log shows this:
>
> Apr 26
to first one that
> answer the call.
Sure. They need separate userids in your sip/iax/zap/whatever.conf so
asterisk knows that they're two separate devices. Then in your dialplan
for that extension just tell it to ring both (with an & between them).
Dial(SIP/device1&
Tony Mountifield wrote on 4/26/07 12:26 PM:
> In article <[EMAIL PROTECTED]>,
> Dave Miller <[EMAIL PROTECTED]> wrote:
>> We upgraded our asterisk server to 1.2.18 last night to pick up the
>> security update. Since then, any calls coming in on IAX2 links get
>
out with one of those web-based
systems that named them all with the extension number, so when we
started needing additional devices for the same extension (like
softphones) we just started tacking suffixes on them. SIP/204 and
SIP/204soft for example.
--
Dave Miller
quot; just like the error says. :)
You do have a class named "default" in the config snippet you pasted, so
MusicOnHold(default) should work.
--
Dave Miller http://www.justdave.net/
System Administrator, Mozilla Corporation http://www.mozilla.com
will pick up both. Just be
forewarned, the T1/E1 channels will all get numbered before the POTS
channels, no matter what order they're on the bus, so 1-24 will be your
T1 and 25-28 the POTS, for example. (I think E1 goes to 32?)
--
Dave Miller http://w
gt; asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing l
he link to download the sound files is dead (wyoming.e-tools.com is
> NXDOMAIN).
> Anyone have a copy of these?
I believe they're included in Asterisk's "extra sounds" package now.
Look for the sounds with a "tt-" prefix on the filenames.
--
Dave Miller
Adam Jacob Muller wrote on 5/5/07 1:38 PM:
>
> On May 5, 2007, at 1:15 PM, Dave Miller wrote:
>
>> Adam Jacob Muller wrote on 5/5/07 1:06 PM:
>>> Hi,
>>> I have some annoying telemarketer calling me on a recurring basis, but
>>> I'd like to disco
extra blank line at the end:
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: from-inside (or whatever context is appropriate)
Exten: (the number you want to call)
Priority: 1
I'm going from memory, so you may have to play with it a little bit but
that&#x
cannot contact the boot server when the other
> 2 can?
Have you checked their boot server type, and does it match what you have
available? If FTP is all you have set up on the boot server and those
two phones are set to use TFTP then you would have this issue.
--
Dave Miller
ny more suggestions.
>
> Steve
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Miller
> Sent: Sunday, 27 May 2007 5:56 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] reset Po
22 matches
Mail list logo
