Displaying 20 results from an estimated 5000 matches similar to: "Using a 1.4 config with 1.6"
2008 Nov 05
5
Phishing attempt
FYI/Heads up,
I /just/ received what looks like a phishing attempt for information
about Open Source PBX usage. It says it comes from Digium but all the
links (including the one for digium.com) point elsewhere.
Rod
--
2009 Feb 03
1
Authentication woes.
I'm still searching but hoping someone can offer a clue-stick.
Long story short! I had a server crash suddenly and all I can get at
are the files. Built a new host and copied the data and config files
over, correcting ownership and permissions (hopefully) as I went.
But now I can't get logged in.
Messages in /var/log/dovecot/dovecot-info.log, without saslauthd
running, are like
2008 May 12
2
Which sound file formats?
I've got the text files created -- thanks to Russell Bryant -- for
re-building the core and extra sounds using another voice but I'm not
sure which formats to actually build.
This will be a small/personal system using Vitelity.net so will only
have SIP connections.
The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729,
.gsm, .ulaw, and .wav.
What are the minimal
2010 Aug 31
4
No audio on call forward after upgrade from Asterisk 1.4 to 1.6
Hi everyone,
This is my first post to the list, although I am a long term user of Asterisk. I have recently found a problem that I just can't seem to solve.
I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have one single
2010 Aug 24
4
1.6 and asterisk gui
Hello,
I'm new to asterisk and this list. The ISO download appears to have
1.6 with the FreePBX GUI but I am looking to use the Asterisk GUI.
The only option for the Asterisk GUI is to use 1.4. Is it as simple
as installing 1.6 only then using the yum repository to install the
Asterisk GUI? If so, what packages are needed?
Thanks!
2008 Oct 07
2
Virtual domain aliases
As I said in a previous reply the server is going great. In fact I can
even send mail via it. (On the really old server I'm moving from I
couldn't get authentication for outbound to work.)
I now have a couple of small issues to deal with before moving
completely off the old system.
Virtual domains aliases? My reading seems to indicate that Postfix only
handles aliases in one
2010 Nov 12
3
Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Hi All,
I'm having an issue where Asterisk continuously sends out a GAZILLION "SIP NOTIFY" messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine.
I'd like to confirm the layout of the
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
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2008 May 07
3
[Q] IMAP update message header?
More about IMAP, less about Dovecot.
This is more an IMAP question than specifically about Dovecot. Sorry to
ask here but I've been researching, reading and lurking with no answer
so far.
What I'm doing is running reports as a user from a remote system to
provide a list of messages (Sender/From, Subject, Date, size) in a
specific folder(?). In one case it isn't even a Dovecot
2008 Oct 08
1
Dovecot-sieve processing optimizations
I'm working at the next part of the virtual domains mail server. I'm
moving this account (raanders at acm.org is a forwarder) which has a bunch
of procmail rules to file into folders.
My question is if it is more efficient is use?
if {
...
} elsif {
...
} elsif {
...
}
else
This seems to be the way many of the example scripts do it but I found
at least one that used
if
2010 Oct 21
10
Asterisk 1.80-rc5
Just done a clean install of rc5 on a totally new machine and found the
following:-
/etc/init.d/asterisk start
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
Dave Cotton
2010 Jul 27
1
Asterisk and Amazon Web Services
Anyone tried installing Asterisk in a AWS server?
\\||/
Rod
--
2010 Oct 14
6
Audiocodes firmware
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body text="#000000" bgcolor="#ffffff">
<font size="+1">Does anyone have links to the most recent audiocodes
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
2010 Oct 22
5
dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook?
2010 Sep 14
9
Speech To Text on linux with asterisk
Hi,
Is it possible to record say 30 seconds of audio and then have LumenVox
convert to text ?
or any available tool open source for speech to text .
Regards
Dhaval
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2010 Jul 24
4
getting some segmentation faults with 1.8
I downloaded the latest 1.8 (27922) but got some segmentation faults.
The first one was when it loaded cdr_odb, and so I changed menuselect
not to compile that one, but the second one was when it tried to load
chan_agent and so I stopped there to see if anyone else was seeing
this. The agents.conf is all commented out except for [general] .
Anyone know what is happening?
Thanks.
P.S. I deleted
2010 Sep 11
8
No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Hi Everyone,
I have a provider whose DID used to come into the box just fine but recently
stopped working. Nothing has been changed on our end.
Here is what I get when doing "sip set debug peer PROVIDER":
Sending to 123.123.123.123 : 5060 (no NAT)
^^^^ That is ALL I am getting with sip debug turned on.
With Allow Anonymous SIP set to YES, then the call comes in properly and you
see
2010 Oct 25
1
dovecot Digest, Vol 90, Issue 102
> From: "Roderick A. Anderson" <raanders at cyber-office.net>
> Subject: [Dovecot] RHEL5/CentOS5 YUM repo, rpm, or spec file for 2.0?
>
> I don't remember sing any mention come across the list reference the
> Subject line and nothing shows up within the first three pages of a
> Google search.
>
> Anyone know of a YUM repo. RPM or spec file for