Guys,
Some newbies have no idea. They need to be politely directed to sources
of information. Not left hanging in the air. For obvious reasons, my
answer is incomplete.
On Sun, Mar 20, 2011 at 01:33:34PM +0530, Kaushal Shriyan wrote:
Hi,
I have couple of questions regarding Asterisk.
[18884732963@from-fax-machine:... - your call is hitting the
from-fax-machine context - yet your 'fax' exten is in the from-pstn-4
context. See the [2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c:
Fax detected, but no fax extension line.
When Asterisk detects an incoming fax tone - it tries to
Hello,
I installed 1.6.2.17 version of asterisk.
Set the user database to realtime.
I have no problems with sip users.
They can register talk etc..
With iax clients, they can register also.. And when they call iax to sip, it
works.
When they make an echo test..no voice received on iax clients.
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
it returns 20110321-034329
but it is exactly 8 hours early than the system time when I type date in
linux terminal
Mon Mar 21 19:43:35 HKT 2011
I am looking for help.
CK
what do you get when you do
${STRFTIME(${EPOCH},,%G%m%d-%H%M%S)}
?
Ish
On Mon, 2011-03-21 at 19:45 +0800, asterisk asterisk wrote:
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
it returns 20110321-034329
but it is exactly 8 hours
asterisk asterisk wrote:
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
I don't specify timezone when I'm working with EPOCH, what happens when
you remove it?
Doug
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Ben Franklin quote:
Those who would give up Essential Liberty to
aster...@ck-lee.com wrote:
It is about the same.
Then my guess is that the machine's timezone is incorrect.
Doug
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Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
The timezone is correct. I have double checked.
From CK Lee 李卓廣
On 21 Mar, 2011, at 9:10 PM, Doug Lytle supp...@drdos.info wrote:
aster...@ck-lee.com wrote:
It is about the same.
Then my guess is that the machine's timezone is incorrect.
Doug
--
Ben Franklin quote:
Those
On Mon, Mar 21, 2011 at 09:23:29PM +0800, aster...@ck-lee.com wrote:
The timezone is correct. I have double checked.
How did you check it? What did you check, specifically?
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Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
With gmt+8, the result is
-Mon Mar 21 13:47:59 2011
For linux server timezone I set it via webmin and /etc/localtime is my
timezone file i.e. HK at GMT+8
On Mon, Mar 21, 2011 at 9:36 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, Mar 21, 2011 at 09:23:29PM +0800,
On Mon, Mar 21, 2011 at 07:45:37PM +0800, asterisk asterisk wrote:
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
it returns 20110321-034329
but it is exactly 8 hours early than the system time when I type date in
linux terminal
Mon
it returns 20110321-034329
but it is exactly 8 hours early than the system time when I type date in
linux terminal
Mon Mar 21 19:43:35 HKT 2011
Have you tried ${STRFTIME(${EPOCH},Hongkong,%G%m%d-%H%M%S)} ?
$ date ; TZ=UTC date ; TZ=Hongkong date
Mon Mar 21 10:13:31 EDT 2011
Mon Mar 21
FreeSWITCH
On Mon, Mar 21, 2011 at 4:23 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
Guys,
Some newbies have no idea. They need to be politely directed to sources
of information. Not left hanging in the air. For obvious reasons, my
answer is incomplete.
On Sun, Mar 20, 2011 at
Hi,
I have installed Asterisk 1.8 and am using realtime dialplan contexts from a
mysql table.
Asterisk keeps segfaulting. When I trace the thread ids associated with the
segfaults in the full log, all they have in common is
netsock2.c: == Using SIP RTP CoS mark 5
which is probably a red
Hey Guys,
I knew this is stupid question but i just want to know what is the difference
between Queue member logged out vs Pause ?
-Satish
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Hi,
This may be related to an issue I added to the bug tracker. Problems around
using Local Channels across realtime / non-realtime contexts in 1.8.
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Naomi
On 11-03-21 12:22 PM, Naomi Rosenberg wrote:
When compiling, I ticked all the addons that mentioned mysql, as well as the
DON'T OPTIMIZE flag, but other than that I left it to the defaults.
I would really like some ideas about what might be causing this. If you need
any more information
Satish,
Paused is like Not Ready on other systems. The user is logged in but is
working on something else or took a break (e.g. restroom). Calls rotate pass
the user while paused.
v/r,
Louis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Thanks Louis,
We have only single queue. so is it ok if we use only pause option not logged
out ?
-Satish
From: carreir...@gmail.com
To: asterisk-users@lists.digium.com
Date: Mon, 21 Mar 2011 12:36:09 -0400
Subject: Re: [asterisk-users] Queue pause vs logged out ?
Satish,
Paused
Thanks for your replies. I will try generating a backtrace. I guess it's time
for me to dip my toe into gdb.
Regarding the known issue - I am not using local channels; in fact there are no
local extensions. When you dial in to a DDI it dials straight out again over a
SIP or IAX trunk.
Also,
On 11-03-21 12:47 PM, Naomi Rosenberg wrote:
Thanks for your replies. I will try generating a backtrace. I guess it's time
for me to dip my toe into gdb.
Regarding the known issue - I am not using local channels; in fact there are no
local extensions. When you dial in to a DDI it dials
I don't think there is a functional difference when it comes to which phone
is ringing. But the status appears as paused, as opposed to the user not
being in the queue.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish
We'd need to see the backtrace, but possible.
Is this any use?
#0 0x7f61ae1dfb52 in ?? () from /lib/libc.so.6
#1 0x004f88dd in ast_sockaddr_resolve (addrs=0x7f618ce91770, str=0x0,
flags=0, family=2) at netsock2.c:235
#2 0x7f6197c2be9a in ast_sockaddr_resolve_first_af
Greeting Users List,
Just a friendly reminder that the first Asterisk Tech Tips webinar will take
place this coming Thursday at 12PM. The first half will be an in-depth tutorial
on calling with Google presented by Malcolm Davenport, Senior Product Manager
for Asterisk. Malcolm will teach
I'd suggest try recording in ulaw first and then convert all to wav after.
It may have something to do with timing since you're using iax, what are you
using as a timing source? Hardware or software?
Sent from my iPhone--
_
On Monday 21 March 2011 06:45:37 asterisk asterisk wrote:
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
it returns 20110321-034329
but it is exactly 8 hours early than the system time when I type date in
linux terminal
Mon Mar 21 19
I believe I am using software, since I don't have any special hardware
installed. I do have a Digium card somewhere in a box. I can try
recording in ULAW, but I'm unsure how to convert it to WAV. I think
Asterisk can do it, so I'll give it a try and look it up.
If I can get this sorted out, I
We are glad to announce that ITSPtec now offers a complete
ITSP system for Asterisk with powerful routing engine, billing System-
including invoicing, configuration, phone auto-provisioning and tones of other
features.
For Asterisk, we offer a Multi-Tenant Hosted PBX system with Reseller
I just finished a fresh install of 1.8.3.2 at home using the packages
Digium hosts.
After correcting a number of typo/config'o error that had crept in
over the years, I thought I had everything working.
My wife just complained that she cannot call her mother (who is using an
old IAX hardphone I
damn, advertisements everywhere, also in non commercial mailing lists...
ITSPTEC.COM seems don't understand what a NON-COMMERCIAL DISCUSSION is
about
I will never buy anything from people like you who don't seems to
understand so basic things
@itsptec.com should be blacklisted...
On
Thanks John Bower and ITSPTEC.COM, you've made it easy for me to not feel
bad about never using your products...
On Mon, Mar 21, 2011 at 7:45 PM, Juan hardwareven...@gmail.com wrote:
damn, advertisements everywhere, also in non commercial mailing lists...
ITSPTEC.COM seems don't understand
Hello List,
I have few installations out there based on 1.6.1 or above.
I'm trying to play different voice mail messages based on certain criteria's.
For example, I want during office hours to play (in short): we are not
available to take your call, please leave a message, during off-hours and
great way to kill sales for your company idiot.!
On Mon, Mar 21, 2011 at 7:38 PM, john.bo...@itsptec.com wrote:
We are glad to announce that ITSPtec now offers a complete ITSP system for
Asterisk with powerful routing engine, billing System- including invoicing,
configuration, phone
Even worse... now it smells of a scam
Registered through: GoDaddy.com, Inc. (http://www.godaddy.com)
Domain Name: ITSPTEC.COM
Created on: 27-Jan-11
Expires on: 27-Jan-12
Last Updated on: 27-Jan-11
On Mon, Mar 21, 2011 at 9:06 PM, Outback Dingo
Oh my...
On Mon, Mar 21, 2011 at 8:09 PM, Outback Dingo outbackdi...@gmail.comwrote:
Even worse... now it smells of a scam
Registered through: GoDaddy.com, Inc. (http://www.godaddy.com)
Domain Name: ITSPTEC.COM
Created on: 27-Jan-11
Expires on: 27-Jan-12
Last
I listened to your email using DriveCarefully and will respond as soon as I can.
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While you're testing, capture iax2 debug info as well as that may point to
other factors/issues.
Possible fixes:
File conversion:
Yes asterisk can do the conversion
File convert xyz.ulaw xyz.wav
As long as you selected wav format in the initial build.
Timing module for iax:
Check your
On Mon, Mar 21, 2011 at 8:05 PM, Harel Cohen ha...@easycall.gi wrote:
Hello List,
I have few installations out there based on 1.6.1 or above.
I’m trying to play different voice mail messages based on certain
criteria’s. For example, I want during office hours to play (in short): “we
are
Hi all
In asterisk source code we can see lots of places
AST_CDR_FLAG_LOCKED flags is used.This is for CDR purpose. Does anyone
what is exact usage of this lock in CDR.If I remove this flags where it
will impact,any data overwrite will happen..?
Thanks
Nikhil
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