Hello list,
my CLI is spammed with :
[Sep 13 08:31:38] doing dnsmgr_lookup for 'ssw6.itsp.tld'
[Sep 13 08:31:38] doing dnsmgr_lookup for 'ssw6.itsp.tld'
[Sep 13 08:31:47] doing dnsmgr_lookup for 'ssw4.itsp.tld'
[Sep 13 08:31:48] doing dnsmgr_lookup for 'ssw4.itsp.tld'
[Sep 13 08:31:49]
On Sun, 12 Sep 2010, Kevin Keane wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Sunday, September 12, 2010 5:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Sun, 2010-09-12 at 15:32 -0700, Kevin Keane wrote:
In terms of telephony, a T-1 can make a huge difference over DSL. DSL
gives you a lot of raw bandwidth, true, but for voice that really
doesn’t matter all that much. Voice calls only take a relatively small
amount of bandwidth anyway; you
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Sunday, September 12, 2010 11:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Moving from
Hello list,
what is the correct syntax ?
exten = s,n,Queue(${queuename}${timeout},cleanpickup.agi^${CHANNEL})
[Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to
execute
'/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-017a':
File does not exist.
Hi,
Searching this list archives, I couldn't find a definitive answer to my
question :
how to send SMS to Gigaset phones ?
My goal is to send Alert SMS such as This phone system will be stopped in
5mn for maintenance to every terminal (SIP phones and Gigaset DECT phones).
(So at the moment, I'm
On 11 September 2010 20:36, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
On 09/09/10 17:59, Steve Davies wrote:
On 9 September 2010 17:52, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Steve Davies wrote:
Hi,
I am using 1.6.2.11, and I need to be able to include the name
Hi,
We have a user who is putting large call volumes through Asterisk, and
wants to BLF monitor up to 90 extensions. We are struggling to find a
handset that can keep up with Asterisk :)
1) Is there a handset that will do this?
2) Is there a different (standard) way to send BLF and allow
Hi,
In voicemail.conf you can choose among several file format (wav, wav49 and
gsm) with which voicemail are saved.
Which one the is the most widely read by Windows, Mac and Linux PC media
players ?
Suggestions ?
Regards
--
_
On 09/13/2010 10:11 AM, Steve Davies wrote:
On 11 September 2010 20:36, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
On 09/09/10 17:59, Steve Davies wrote:
On 9 September 2010 17:52, Antonio Berrios
anto...@sheffieldcitytaxis.comwrote:
Steve Davies wrote:
Hi,
I am using
On 13 September 2010 11:07, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Gotcha. Yeah, I'm looking at implementing that (searching call
recordings by agent that took the call) here but since our asterisk call
recording is a separate server to the ones dealing with queues I'll be
This is a problem with extconfig.conf - not your res_ or cdr_ ones.
In your case - extconfig.conf probably contained something like
'sippeers = mysql,MyDBase,sippeers'. The 'problem' is that the middle
parameter is no longer for the database name - it is for the context in
res_mysql.conf. So,
We have good success with Grandstream, but even though we monitor
100 phones, I don't think we get the amount of NOTIFY messages
you describe. But you should try them.
Ron
Op 13-09-10 11:56, Steve Davies schreef:
Hi,
We have a user who is putting large call volumes through Asterisk, and
Hi,
In voicemail.conf.sample, you can read this:
format=wav49|gsm|wav
; WARNING:
; If you change the list of formats that you record voicemail in
; when you have mailboxes that contain messages, you _MUST_ absolutely
; manually go through those mailboxes and convert/delete/add the
; the message
2010/9/13 Steve Davies davies...@gmail.com
Hi,
We have a user who is putting large call volumes through Asterisk, and
wants to BLF monitor up to 90 extensions. We are struggling to find a
handset that can keep up with Asterisk :)
1) Is there a handset that will do this?
2) Is there a
Hi,
On 09/13/2010 10:57 AM, Olivier wrote:
Hi,
In voicemail.conf you can choose among several file format (wav, wav49
and gsm) with which voicemail are saved.
Which one the is the most widely read by Windows, Mac and Linux PC media
players ?
I use wav49. It is compatible with Windows and
On 13 September 2010 11:43, Olivier oza_4...@yahoo.fr wrote:
2010/9/13 Steve Davies davies...@gmail.com
[snip]
Our test involves about 10 BLF-NOTIFY messages per second to each
handset with a 5-second pause every 5 seconds. This will either crash
or render unusable all of the following
Hi,
On 09/13/2010 11:34 AM, Olivier wrote:
Hi,
In voicemail.conf.sample, you can read this:
format=wav49|gsm|wav
; WARNING:
; If you change the list of formats that you record voicemail in
; when you have mailboxes that contain messages, you _MUST_ absolutely
; manually go through those
Hello,
Am 13.09.10 11:56, schrieb Steve Davies:
Hi,
We have a user who is putting large call volumes through Asterisk, and
wants to BLF monitor up to 90 extensions. We are struggling to find a
handset that can keep up with Asterisk :)
1) Is there a handset that will do this?
we only use
On 13 September 2010 12:16, Stefan Schmidt s...@sil.at wrote:
Hello,
Am 13.09.10 11:56, schrieb Steve Davies:
Hi,
We have a user who is putting large call volumes through Asterisk, and
wants to BLF monitor up to 90 extensions. We are struggling to find a
handset that can keep up with
As a side note to this - do NOT try and use Aastra's - as they tend to
crash after 50 BLF's!
Also, could you please send me (perhaps off-list to a...@datavox.co.uk)
your Yealink T28 findings - as I am a beta tester for them?
Cheers
Andy
-Original Message-
From:
On 13/09/10 11:03 PM, Steve Davies wrote:
On 13 September 2010 11:43, Olivieroza_4...@yahoo.fr wrote:
2010/9/13 Steve Daviesdavies...@gmail.com
[snip]
Our test involves about 10 BLF-NOTIFY messages per second to each
handset with a 5-second pause every 5 seconds. This will either crash
or
On Mon, 13 Sep 2010, Olivier wrote:
Hi,
Searching this list archives, I couldn't find a definitive answer to my
question :
how to send SMS to Gigaset phones ?
My goal is to send Alert SMS such as This phone system will be stopped in
5mn for maintenance to every terminal (SIP phones and
On Mon, Sep 13, 2010 at 1:32 AM, Kevin Keane subscript...@kkeane.comwrote:
My numbers are from an ATT DSL line in California, suburban San Diego
county, and just around the corner from the central office. So it is not the
distance (with DSL, the distance does make quite a difference). On the
On Mon, Sep 13, 2010 at 3:29 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Mon, 13 Sep 2010, Olivier wrote:
how to send SMS to Gigaset phones ?
I dimly recall someone doing this and publishing a page of script... but where?
Look using Google back from 2004-2008 something like sms
2010/9/13 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Mon, 13 Sep 2010, Olivier wrote:
Hi,
Searching this list archives, I couldn't find a definitive answer to my
question :
how to send SMS to Gigaset phones ?
My goal is to send Alert SMS such as
Christian Weeks c...@weeksfamily.ca writes:
Hello
I purchased an AEX800 card to replace the ageing cheap channel bank/T1
card solution a few months ago, assuming that it would be a more robust
solution for my small scale phone system. However, it appears to be
anything but that.
Originally
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
I have been having issues with hackers doing registration scanning against
our server. We block their address at the fire wall but since asterisk does
not force a drop of the connect after so many bad reg
On Monday 13 September 2010 06:07:09 Sebastian wrote:
On 09/13/2010 11:34 AM, Olivier wrote:
In voicemail.conf.sample, you can read this:
format=wav49|gsm|wav
; WARNING:
; If you change the list of formats that you record voicemail in
; when you have mailboxes that contain messages,
Steve
Grandstream has a new services GXP-21XX coming out they may work for your.
We have been a beta tester and the BLF on these seem to work much better
then the GXP-20XX units. I do not have the side cars in stock right now so
I don't know how they work with it but you can put at least two
On Mon, Sep 13, 2010 at 11:22 AM, Bryant Zimmerman brya...@zktech.com wrote:
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
Not within Asterisk. Google fail2ban
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com |
On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote:
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
Consider writing a filter for fail2ban [http://www.fail2ban.org/] that
works on the Asterisk logs?
--
On Mon, 2010-09-13 at 12:49 +1200, Matt Riddell wrote:
On 11/09/10 12:44 PM, Carlos Chavez wrote:
The past few days I started having a problem with a small call center
setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam
is
configured to auto answer the call.
As I look to move our systems to version 1.8 I am looking at making a
change from mySQL to PostgreSQL.
I love mySQL but am getting very concerned about i'ts new owners.
Should I be able to move all my realtime stuff to PostgreSQL is it fully
supported with asterisk?
Is there any down side to
Hello,
can anyone please tell me how I can give arguments to my AGI script ?!
I think asterisk sees the name of the AGI + the channel as one filename,
and of course this file then does not exist.
Jonas.
On 09/13/2010 10:26 AM, Jonas Kellens wrote:
Hello list,
what is the correct syntax ?
What strategy are you using for the Queue?
We are using Least Recent at the moment. Why would queue strategy
impact this?
Carlos: I had similar issues, caused by a setting somewhere in the advanced
section of eyeBeam. Something about Disconnect if no audio received for x
On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote:
On 13 September 2010 11:07, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Gotcha. Yeah, I'm looking at implementing that (searching call
recordings by agent that took the call) here but since our asterisk call
recording is a
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, September 13, 2010 10:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] PostgreSQL is asterisk friendly with it?
can you state your internet connection your agents are on?and one more thing..
how are the members positioned into the Queue? static? Dynamic? single station
and call forwarding (find me follow me extension in the queue)? do you get call
waiting override with Auto Answer?
-- Tarek Sawah
On Mon, 2010-09-13 at 17:48 +0200, Jonas Kellens wrote:
Hello,
can anyone please tell me how I can give arguments to my AGI script ?!
I think asterisk sees the name of the AGI + the channel as one
filename, and of course this file then does not exist.
In Asterisk 1.4 you use
On Mon, 13 Sep 2010, Danny Nicholas wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, September 13, 2010 10:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On 13 September 2010 16:58, Carlos Chavez cur...@telecomabmex.com wrote:
On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote:
On 13 September 2010 11:07, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Gotcha. Yeah, I'm looking at implementing that (searching call
recordings by
On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote:
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
I have been having issues with hackers doing registration scanning against
our server. We block their address at the fire wall but since
On Mon, Sep 13, 2010 at 12:31:55PM -0400, Vince Vielhaber wrote:
change from mySQL to PostgreSQL.
I love mySQL but am getting very concerned about i'ts new owners.
Should I be able to move all my realtime stuff to PostgreSQL is it fully
[snippage and probably off topic]
Why are you worried
At 11:38 PM 9/12/2010, you wrote:
my CLI is spammed with :
[Sep 13 08:31:38] doing dnsmgr_lookup for 'ssw6.itsp.tld'
How can I turn this off ?!
I just change the 4 in the source code to 14 so it doesn't show up
till you add a lot of Vs. It only occurs in one place in the code so
it's
I agree with Steve, I've had a couple of people speak with me about
concerns of MySQL's new ownership, but I stand by the same reasons
Steve doesMySQL will not be going anywhere, and if it does,
someone'll just fork and keep it going under a new name.
On Mon, Sep 13, 2010 at 11:46 AM, Steve
Hi Olivier,
I remember having had a similar discussion a few years ago. I will paste
my postings from around May 2007 further down.
First, I did not try sending SMS over VOIP to the phone, just over Voip
to an ATA and then over analogue line (or ISDN) to the phone. So I have
no idea wether the
On Mon, 2010-09-13 at 15:59 +, Tarek Sawah wrote:
can you state your internet connection your agents are on?
and one more thing.. how are the members positioned into the Queue?
static? Dynamic? single station and call forwarding (find me follow me
extension in the queue)? do you get call
Steve
I have 64 channels being monitored with an SPA962 with two SPA932
sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very
happy with this. Latest firmware is a must.
HTH
Cassius Smith
--
_
-- Bandwidth and
On 09/13/2010 06:01 PM, Carlos Chavez wrote:
On Mon, 2010-09-13 at 17:48 +0200, Jonas Kellens wrote:
Hello,
can anyone please tell me how I can give arguments to my AGI script ?!
I think asterisk sees the name of the AGI + the channel as one
filename, and of course this file then does not
Hose hose+aster...@bluemaggottowel.com writes:
The most straightforward way would be to just define explicit patterns.
Obviously that works, but doesn't seem scalable in terms of maintenance.
I don't think that maintaining the list in the dial plan is all that
bad, actually. Dump it in its own
Bryant Zimmerman brya...@zktech.com writes:
As I look to move our systems to version 1.8 I am looking at making a
change from mySQL to PostgreSQL.
I love mySQL but am getting very concerned about i'ts new owners.
Should I be able to move all my realtime stuff to PostgreSQL is it fully
On Mon, Sep 13, 2010 at 08:15:34PM +0200, Jonas Kellens wrote:
[Sep 13 20:14:59] -- Launched AGI Script
/var/lib/asterisk/agi-bin/cleanpickup.agi
[Sep 13 20:14:59] opruimenpickup.agi: Failed to execute
'/var/lib/asterisk/agi-bin/cleanpickup.agi': Permission denied
So check that
No that is not the problem, nor was it the question.
I found the solution. Apparently you need to place the AGI-script and
its arguments between , but the arguments still need to be separated
by a comma.
Example :
exten = s,n,Queue(queuename,myscript.agi,arg1,arg2)
If anyone wants
Hello,
anyone on this list knows how to turn these messages off please ?!
I have in sip.conf :
srvlookup=no
and in dnsmgr.conf :
[general]
enable=no; enable creation of managed DNS lookups
; default is 'no'
;refreshinterval=1200; refresh managed DNS lookups every n
i have this scenario where i have a marketing department calling USA numbers
excessively and sometimes the leads contain duplicate numbers OR duplicate
customers with different numbers on the other hand we have some numbers that
are black listed the destination should be checked and caller
On Mon, 2010-09-13 at 00:32 -0700, Kevin Keane wrote:
Latency also is the reason VoIP does not work at all over satellite
connections even though they tend to have plenty of bandwidth.
Please define does not work at all over satellite ???
Sure, it is not studio HIFI quality, but is th
On Mon, Sep 13, 2010 at 2:53 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
anyone on this list knows how to turn these messages off please ?!
*CLI core set verbose 0
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet h...@a-domani.nl wrote:
No these are also geo-stationary (same altitude, so same delay),
commercial and military satelites,
Yes, exactly. Geostationary satellites have been used for telephone for
ages (and are still used for remote areas - they
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
'SIP/crocus-ua-0004' refused to negotiate T.38
[Sep 13 00:46:02] WARNING[3283]: app_fax.c:223
On Mon, Sep 13, 2010 at 4:33 PM, Stanislav Korsei kor...@rinogo.com wrote:
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
'SIP/crocus-ua-0004' refused
On 09/13/2010 10:22 AM, Bryant Zimmerman wrote:
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
I have been having issues with hackers doing registration scanning
against our server. We block their address at the fire wall but since
asterisk does not
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Monday, September 13, 2010 12:13 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Moving from DSL to T1
On Mon,
On 09/14/2010 04:33 AM, Stanislav Korsei wrote:
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
Why install 0.0.5? Its ancient. the world has moved on.
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
On 09/14/2010 04:23 AM, Joel Maslak wrote:
On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet h...@a-domani.nl
mailto:h...@a-domani.nl wrote:
No these are also geo-stationary (same altitude, so same delay),
commercial and military satelites,
Yes, exactly. Geostationary satellites
Hi all,
I would like to install asterisk as my home pbx, Anyone can suggest
which sub version of 1.6 is stable?
Thanks
Nikhil
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Your string and boot-option look good.
In the SonicWall config, its a two step process:
- create the new boot option under DHCP Server menu Advanced button
Add Option
- assign it to your lease scope under DHCP Server menu DHCP Server Lease
Scopes section Edit button Advanced tab DHCP
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
--
_
-- Bandwidth and Colocation Provided by
On Tue, Sep 14, 2010 at 1:01 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Is it possible to record say 30 seconds of audio and then have LumenVox
convert to text ?
ASR, yes.
http://www.digium.com/en/products/software/lumenvox.php
--
Paul Belanger | dCAP
Polybeacon | Consultant
Thanks Paul,
i think still i have some problem to understand , i mean to say that i have
30 minutes audio file in
WAV format and i wnat its text here are the scenario .
- Call comes in
- start recording
- call remains for 30 minutes
- stop recording
- convert wav file audio to text.
is this
It is simply not possible, though it might be in the distant future.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-14 1:50 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote:
Thanks Paul,
i think still i have some problem to understand , i mean to say that i have
30 minutes audio file
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