Hi Tilghman , thanks for the help.
ControlPlayback can't be used with ExternalIVR, can it ?
We use ControlPlayback in our current dialplan, what I am wanting (in
concept) is to have a meetme/conference room where one of the parties
is a caller, and the other party a file to be controlplaybacked
Is it possible to send sip messages (debug) to a file or syslog server
without having them present in the console? If so, does anyone know what
kind of performance hit this would create.
Instance has approx 800 sip peers.
--
_
hi,Dear all
as you know, MeetMe has cdr for each attendant. but the fee is
always paid by the moderator. not by each one.
and the the members in the conference are dynamic changed.
in this scenario, how to billing for MeetMe conference? and i want to
hungup all the calls when the account
Thank you ! Do you konw how to realtime billing for MeetMe conference?
2010/8/30 Paul Belanger paul.belan...@polybeacon.com:
On Sun, Aug 29, 2010 at 10:52 PM, Zhang Shukun bit...@gmail.com wrote:
but i want to know if i can invite some one to the conference when i
already in the conference?
Hi,
I've recently had a fairly prolonged SIP registration attack, 18 hours in
this case and often with 200 attempts per second, and suspect I've had a
number of these in the past. The main symptom I noticed previously was,
because Asterisk was responding to each registration request it
- Flavio Miranda flaviormira...@hotmail.com wrote:
Hi all,
I am trying to set up Web-meetme in Asterisk 1.6. After some attemps, I am
receiving the message: DB Error: connect failed
What could be ?
It's very likely the connection failed to your database... /sarcasm
Check
On Mon, 30 Aug 2010, Nikhil Nair wrote:
Hi,
I've recently had a fairly prolonged SIP registration attack, 18 hours in
this case and often with 200 attempts per second, and suspect I've had a
number of these in the past.
Almost everyone has - read the fine archives, then google for
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the
On Mon, Aug 30, 2010 at 3:19 AM, Positively Optimistic
positivelyoptimis...@gmail.com wrote:
Is it possible to send sip messages (debug) to a file or syslog server
without having them present in the console?
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com wrote:
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
Post a debug log of the problem:
Todd
How do you have the context in the phones sip configs set?
Bryant
From: Todd Reese trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on
Here is the sip.conf portion for extension 150
[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device 150
accountcode=
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
Sent: Monday, August 30, 2010 2:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Play a number
Hello,
Is there any option to set the maximum number of seconds a caller can wait
in a queue before being pulled out ?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
On 08/30/2010 01:58 AM, Julian Lyndon-Smith wrote:
ControlPlayback can't be used with ExternalIVR, can it ?
We use ControlPlayback in our current dialplan, what I am wanting (in
concept) is to have a meetme/conference room where one of the parties
is a caller, and the other party a file to
Here's a debug for extension 150
[Aug 30 11:34:53] VERBOSE[2099] config.c: == Parsing
'/etc/asterisk/logger.conf': [Aug 30 11:34:53] DEBUG[2099] config.c:
Parsing /etc/asterisk/logger.conf
[Aug 30 11:34:53] VERBOSE[2099] config.c: == Found
[Aug 30 11:34:53] VERBOSE[2099] logger.c:
Todd
Your context must be set to where you want your extension to start each
time it dials out. Without getting into your dialplan code too much try
changing the context to point to dialout1
context=dialout1
If dialout1 is working you should be able to dial.
The best way to handle this is to
It is a half turned up PRI, so 1-12 should be correct?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: August 28, 2010 12:49 PM
To: asterisk-users@lists.digium.com
Subject: Re:
I tried those as you said, deleting my failed attempt. I've found that
using hardhdlc=24 generates an error and reminds me that FXO uses FXS
signalling and vice versa when running dadhi_restart, which seems to
indicate that it is the wrong variable name.
I also notice that if I change that
On Mon, Aug 30, 2010 at 10:31 AM, Tino t...@sparksupport.com wrote:
Hello,
Is there any option to set the maximum number of seconds a caller can wait
in a queue before being pulled out ?
Thanks
In the Queue() command itself there is a timeout parameter. From your
asterisk box, try
Thanks Warren for your help
On Mon, Aug 30, 2010 at 9:21 PM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Aug 30, 2010 at 10:31 AM, Tino t...@sparksupport.com wrote:
Hello,
Is there any option to set the maximum number of seconds a caller can wait
in a queue before being pulled out ?
Gordon Henderson wrote:
So.. Get a copy of the sipvicious code from http://blog.sipvicious.org/
(or directly from http://code.google.com/p/sipvicious/ ) and learn how to
use svcrash.py as that's the only thing that's going to ultimately stop a
long-term attack on your site. For now,
Unfortunately, that didn't work. The phone is still giving me a 404
error.
I have my own system that is 1.6.2.7 with Grandstream phones that works
fine. Using it as a guide, I built this server for a client which also
has Grandstream phones.
Last week, it dialed out fine. Since the
In my old office, for conference purpose , gotomeeting was used. also for
the lecture delivery, same gobomeeting was used, most the time , we need to
listen voice only. also, we use to share desktop screen.
But as far as I know SIP is the standard for video telephony. SIP can handle
video +Audio.
On Mon, Aug 30, 2010 at 11:42 AM, Todd Reese trees...@gmail.com wrote:
Here's a debug for extension 150
In the future, simply attach your debug log to your email. Here is
your problem:
[Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from '150' to extension
'6789542133' rejected because
possibly check you spelling: [from-interal] - [dialout1]
include = from-internal
??
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com wrote:
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My
I actually found that one and corrected it. I have replaced the
context with the from-internal, remote, and dialout1. Each has produced
the same results of a 404 error.
On 8/30/2010 2:10 PM, Paul Belanger wrote:
On Mon, Aug 30, 2010 at 11:42 AM, Todd Reesetrees...@gmail.com wrote:
On 8/30/2010 11:42 AM, jeremy.hellst...@synovate.com wrote:
I tried those as you said, deleting my failed attempt. I've found that
using hardhdlc=24 generates an error and reminds me that FXO uses FXS
signalling and vice versa when running dadhi_restart, which seems to
indicate that it is the
Thanks for pointing out the misspelling. I've corrected that and
still no luck.
On 8/30/2010 2:33 PM, Alex Bell wrote:
possibly check you spelling: [from-interal] - [dialout1]
include = from-internal
??
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com
On Mon, Aug 30, 2010 at 2:48 PM, Todd Reese trees...@gmail.com wrote:
Thanks for pointing out the misspelling. I've corrected that and still no
luck.
Create a new debug log with your recent changes, re-attach it the list.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
The specific error message is as follows.
_
Changing signalling on channel 24 from Unused to Hardware assisted
D-channel
DAHDI_CHANCONFIG failed on channel 24: Invalid argument (22)
Did you forget that FXS interfaces are configured with
On Mon, 30 Aug 2010, J. Oquendo wrote:
How about a little cron script without having to install anything? You
could run it off the hour:
rightnow=`date +%Y-%m-%d %k`
grep $rightnow /var/log/asterisk/messages |\
awk '/No matching peer/' | sed's:'\''::g' |\
uniq | awk '{print iptables -A
On 8/30/2010 2:59 PM, jeremy.hellst...@synovate.com wrote:
The specific error message is as follows.
_
Changing signalling on channel 24 from Unused to Hardware assisted
D-channel
DAHDI_CHANCONFIG failed on channel 24: Invalid
Strange issue that I can't figure out and I am hoping someone may have some
ideas. Two Asterisk boxes running 1.2.34 (yeah I know it is old, but it
runs like a top and I am not going to mess with it). *B rsyncs config from
*A. *A dies. I bring up *B and it all works fine, except for one issue.
Gordon Henderson wrote:
On Mon, 30 Aug 2010, J. Oquendo wrote:
I also posted a very effective iptables script some weeks ago if you care
to search the archives. It works and is extremely effective in blocking
these types of attacks - however, it will not stop a broken sipvicious
from
On 10-08-30 01:53 PM, J. Oquendo wrote:
Gordon Henderson wrote:
On Mon, 30 Aug 2010, J. Oquendo wrote:
I also posted a very effective iptables script some weeks ago if you care
to search the archives. It works and is extremely effective in blocking
these types of attacks - however,
Hi,
I am having a weird issue with a Cisco 9971 phone. I managed to get most of
it working, including the side car, however one of the issues is that there
seems to be some sort of side tone / beep occurring roughly every 13 seconds
or so, as if the phone is activated with call waiting.
However
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