On Sat, Sep 24, 2011 at 9:35 PM, Bruce B bruceb...@gmail.com wrote:
Hi everyone,
I don't mean to be rude but honestly which genius comes up with changing the
Word to the wise -- if one starts a sentence with I don't mean to
be...X your true intentions are to be just that.
If you find yourself
On Thu, Jul 28, 2011 at 4:46 AM, Duncan Turnbull dun...@e-simple.co.nzwrote:
On 28/07/2011, at 8:41 PM, Paul Hayes p...@provu.co.uk wrote:
On 28/07/11 02:58, Mike Diehl wrote:
Any ideas?
Mike.
I'd go on site if possible and see what actually happens at 19:00. Set
up a wireshark
On Thu, Jul 21, 2011 at 7:13 PM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I've got a strange problem with a customer's phones.
They've got a bunch of Grandstreams that seem to be rock solid... until
7:00pm. At 7:00, some of the phones become unavailable, and stay down.
Call
quality
On Mon, Jul 18, 2011 at 9:20 AM, Gilles codecompl...@free.fr wrote:
Hello,
I'd like to run Asterisk on an embedded device, where space is scarce.
It should be able to handle calls from a VoIP provider in SIP, calls
from the PSTN through Dahdi, and voicemail.
If someone's already done this,
On Fri, Jul 15, 2011 at 12:47 PM, CDR vene...@gmail.com wrote:
I need to keep out all connection from 5 countries, which originate
most of the Denial of Service attacks. The entries are
around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter
way to do this by using User Tables in
On Wed, Jul 13, 2011 at 5:35 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
We're using asterisk 1.8.3.2 and are finding incidences of stale
channels remaining after both parties have hung up. We have tried to
hang the channel up using
channel request hangup
But by it's definition, it
On Tue, Jun 21, 2011 at 4:12 AM, randulo rand...@randulo.com wrote:
On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov
abalas...@evaristesys.com wrote:
I nominate this for most imaginative use of Asterisk-users of 2011.
It's already qualified to win in the grammar and spelling categories.
/r
On Mon, Jun 6, 2011 at 11:55 AM, Silver Thorne szilvertho...@gmail.com wrote:
Hello Folks;
Perhaps I am chasing my tail here.
Before I go any further, is this compatible/supported in Asterisk 1.6x? If
so, I would be willing to post any manager.conf or http.conf snippets
needed.
When I
2011/6/2 Örn Arnarson o...@arnarson.net:
To clarify; I observe the exact same results no matter how I connect
to the AMI on this particular server. I tried connecting FROM this
server to an AMI on another server to make sure it wasn't the telnet
client or some such, and then it worked
On Mon, May 30, 2011 at 2:44 AM, gincantalupo
gincantal...@fgasoftware.comwrote:
Hi,
it is a known problem, one of the worst. To avoid it:
- do not use urls, only ip addresses in sip.conf
or put your urls inside /etc/hosts (is what I do especially sip providers
urls)
or install a
On Fri, May 20, 2011 at 4:38 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
If you are going to use call files don't write them directly to
/var/spool/asterisk/outgoing/
write them in some temp directory and then move them to
/var/spool/asterisk/outgoing/
Ish
Make sure that your temp
On Fri, May 20, 2011 at 2:10 PM, satish patel satish...@hotmail.com wrote:
Hi Guys!
This is strange issue with 1.8 I have restarted my asterisk and it destroy
all registered SIP peers now only solution is i manually reboot all phones
to get them register back. I have never seen issue like
On Fri, May 20, 2011 at 3:00 PM, satish patel satish...@hotmail.com wrote:
We have polycom 501 and i am waiting since last 5 min no registration
require appear.
-S
With Polycom 321 you can poke around the menus -- one of them has a
countdown timer which will show you when the next
On Tue, May 17, 2011 at 4:21 PM, satish patel satish...@hotmail.com wrote:
Hey Guys!
Sorry i am posting scripting question in asterisk forum but i had no
choice. also i am not script expert so i though anyone here might help me.
following is my example sip.conf now i want to add
On Sun, May 15, 2011 at 4:08 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
Not exactly. Asterisk is multi-threaded. strae traces a specific thread.
To see the most active thread, press 'H' (shift-h) in top. Wait for the
display to refresh at least twice (on the first time it won't make
On Mon, May 16, 2011 at 10:33 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, May 16, 2011 at 10:01:36AM -0400, Mark Deneen wrote:
strace -f -ff ASTERISK_PID
traces all threads on my system.
But do you really want that?
Asterisk has many threads generating quite a lot
On Fri, May 6, 2011 at 2:14 PM, Jerry Geis ge...@pagestation.com wrote:
Is there a way in asterisk to Activate/Clear the blinking light on polycom
phones
indicating VM. Either from an AGI or some way in the dialplan?
I want to be able to control this light for from my application.
I have an
On Thu, May 5, 2011 at 4:07 PM, Paul Belanger pabelan...@digium.com wrote:
On 11-05-05 12:30 PM, Ira wrote:
At 07:56 AM 5/5/2011, you wrote:
So how can we fix this? How can we get more people involded? What
makes projects like FedoraTesting[3] and DebianTesting[4] popular? How
can the
Satish,
You must register your handle with freenode, because the asterisk
channel only allows registered people in.
http://freenode.net/faq.shtml#nicksetup
-M
On Fri, Apr 29, 2011 at 11:41 AM, satish patel satish...@hotmail.com wrote:
Hey Matt,
I have download irc linux base CLI client and
On Fri, Apr 22, 2011 at 11:02 AM, Paul van der Vlis p...@vandervlis.nl wrote:
Hello,
I cannot call my server over the internet with SIP anymore.
Even when I do a maximum logging on my firewall, I don't see packets
coming from outside. I've tried it from an ekiga.net account and an
Rajnikant,
This surely depends on how you start asterisk. How are you starting
the asterisk process?
-M
On Thu, Apr 21, 2011 at 7:20 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote:
Hi Friend,
Can't get hostname environment variable on asterisk dialplan.
Help me about how to get hostname
On Thu, Apr 21, 2011 at 3:23 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Thu, 21 Apr 2011, RAJNIKANT VANZA wrote:
Can't get hostname environment variable on asterisk dialplan.
1) Is HOSTNAME in the Asterisk process's environment? What does executing:
tr '\000' '\n'
On Thu, Apr 21, 2011 at 4:30 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Thu, 21 Apr 2011, Mark Deneen wrote:
I use runit to manage the asterisk process, and the chpst program
allows fine control over environment and other limits.
runit is intended to be a sysvinit (/sbin/init
On Wed, Apr 20, 2011 at 4:35 PM, satish patel satish...@hotmail.com wrote:
Hey Thanks for that reply after add following option it works but the text
output is totally different.. what its totally different is this dictionary
problem ?
-hmm /var/lib/asterisk/communicator -samprate 8000
In
2011/4/19 Niccolò Belli darkbas...@gmail.com:
Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto:
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and
internet is offline.
srvlookup = no didn't help.
What about putting my provider's name in /etc/hosts?
Should it
dig @193.189.160.13 voip.siol
; DiG 9.6.0-APPLE-P2 @193.189.160.13 voip.siol
; (1 server found)
;; global options: +cmd
;; Got answer:
;; -HEADER- opcode: QUERY, status: REFUSED, id: 35478
;; flags: qr rd; QUERY: 1, ANSWER: 0, AUTHORITY: 0, ADDITIONAL: 0
;; WARNING: recursion requested but not
On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote:
Hey Guys,
Whenever i calling any extension i am getting following WARNING messages do
you have any idea they coming from where?
-Satish
shirley*CLI
== Using SIP RTP CoS mark 5
-- Executing
On Mon, Apr 4, 2011 at 3:20 PM, Jerry Geis ge...@pagestation.com wrote:
I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a
speaker attached.
When asterisk first starts this works. In fact it works for some time. Then
it just stops with this error on the CLI.
[Apr 4
Since my perl skills are pretty much write-only, I've been using
Python. I can actually look at it a year later and immediately see
what the code is doing.
-M
On Fri, Apr 1, 2011 at 9:21 AM, Danny Nicholas da...@debsinc.com wrote:
I'm going to vote for PERL as well. C is not a scripting
Look into the ipt_recent / xt_recent module. It's probably what he is using.
On Wed, Mar 30, 2011 at 4:25 PM, vip killa vipki...@gmail.com wrote:
could you please elaborate on how you have iptables setup to work that way?
On Wed, Mar 30, 2011 at 4:11 PM, Gordon Henderson
On Thu, Mar 24, 2011 at 4:58 PM, Thomas Winter thowin...@googlemail.com wrote:
Hi list,
I have an 44100 Hz file with human voice, stereo with 16Bit.
When convertig this to 8 kHz, mono I loose a lot of quality and have
some ground noise. I tried several sox options but without success.
Can
On Sat, Jan 22, 2011 at 6:53 PM, Carlos Chavez cur...@telecomabmex.com wrote:
On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote
On 22 Jan 2011, at 18:02, Carlos Chavez wrote:
Cannot allocate memory
Have you tried looking at memory?
S
The server has 8gb of ram and 8gb of swap.
On Thu, Jan 20, 2011 at 12:55 PM, Brian C. Huffman
bhuff...@etinternational.com wrote:
Does anyone know how to setup this phone to work with asterisk so that the
indicator light comes on when there's a new message and goes off quickly
(less than a minute) after the message is deleted?
Thanks,
On Wed, Jan 19, 2011 at 2:37 PM, randulo rand...@randulo.com wrote:
Slightly OT: why is the Gmail ad server, which is usually all about
PBX, Asterisk, etc, now showing me Justin Beiber concert tickets on
this thread? Are they seeing it as that childish?
/r
Also OT: Google combines message
On Mon, Jan 17, 2011 at 1:12 PM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Monday, January 17, 2011 11:53 AM
To: Asterisk Users Mailing
On Wed, Jan 12, 2011 at 12:08 PM, Gilles codecompl...@free.fr wrote:
On Tue, 11 Jan 2011 10:02:48 -0500, Mark Deneen mden...@gmail.com
wrote:
Using the shared secret will only allow a single point to point
connection. That is, you have to use certificates if you want more
than one client
On Tue, Jan 11, 2011 at 9:29 AM, Andrew Latham lath...@gmail.com wrote:
On Tue, Jan 11, 2011 at 11:20 AM, Gilles codecompl...@free.fr wrote:
Hello
I read a whole book on OpenVPN, but still can't figure how to
configure the server + client so that the the client connects and
sends SIP/RTP
On Tue, Jan 4, 2011 at 8:52 AM, Andy Graybeal
andy.grayb...@casanueva.com wrote:
On 01/03/2011 07:53 PM, cjwstudios wrote:
Andy,
The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the
2626 can support vlans you can isolate data and voice. Make sure to
spec a UPS on the PoE
Torbjörn,
I don't have experience with asterisk in this regard, but I would
guess that what you have is an elementary stream and not a transport
stream. I believe that VLC could play it back for you.
Outside of that, I would look at using ffmpeg. It may do what you want.
Here is some
Any idea what is it about SIP over IAX2 that made such an improvement?
-M
On Thu, Dec 2, 2010 at 6:01 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
If getting a second circuit is out of the question.
1. Switch to SIP
2. Install and Learn Vyatta for QoS (Squid may help you quite a
But it has a built-in UPS! ;-)
On Tue, Nov 30, 2010 at 10:02 AM, Kyle Kienapfel doctor.w...@gmail.com wrote:
Sounds like they just need to be told its a hilariously bad idea to host
anything important on a cellphone.
On Mon, Nov 29, 2010 at 1:20 PM, Gordon Henderson
On Wed, Nov 24, 2010 at 11:43 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Hello list,
I'm experiencing a lot of server freezes lately. The server just... freezes.
I notice in the log files (/var/log/asterisk/messages /var/log/messages)
that logging stops at the time the server hangs.
On Mon, Nov 22, 2010 at 9:50 AM, Danny Dias ing.diasda...@gmail.com wrote:
2010/11/22 John Novack jnov...@stromberg-carlson.org
Danny Dias wrote:
Hello John,
What i am asking is if i can apply this patch manually or something like
this without making any upgrade of Asterisk, has anyone
On Fri, Nov 19, 2010 at 9:42 AM, Michael voip.quest...@gmail.com wrote:
Hello,
We succeed to send faxes using FFA, when the files are converted to tif
from PDF using gs, but it doesn't work with tif files we copy/upload
directly from our PCs.
We saw in the manual that the size is important,
On Tue, Nov 16, 2010 at 9:28 AM, Gilles codecompl...@free.fr wrote:
Hello
For users who 1) don't have a QoS-capable ADSL router and 2) would
like to run Asterisk with a couple of SIP trunks, I was wondering what
hardware is recommend to run any of the main open-source *WRT projects
to which
On Thu, Nov 18, 2010 at 8:52 AM, Gilles codecompl...@free.fr wrote:
On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneen mden...@gmail.com
wrote:
Are you saying ADSL as in a generic term for broadband router or do
you really mean that the router also acts as a DSL transceiver?
Sorry about
On Thu, Nov 18, 2010 at 9:26 AM, Darrick Hartman
dhart...@djhsolutions.com wrote:
On 11/18/2010 07:52 AM, Gilles wrote:
On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneenmden...@gmail.com
wrote:
Are you saying ADSL as in a generic term for broadband router or do
you really mean that the router
On Mon, Nov 15, 2010 at 3:11 PM, Jeremy Kister
asterisk...@jeremykister.com wrote:
After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP
calls are being destroyed after 1 minute and 20 seconds (80 seconds).
Asterisk is sending a BYE message - I have no idea why.
On Sun, Nov 7, 2010 at 3:58 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
On 11/06/2010 09:18 PM, Sherwood McGowan wrote:
On Sat, Nov 6, 2010 at 2:45 PM, Jonas Kellensjonas.kell...@telenet.be
wrote:
On 11/06/2010 07:18 PM, Tilghman Lesher wrote:
On Saturday 06 November 2010 11:22:06
On Fri, Nov 5, 2010 at 1:18 AM, Bruce B bruceb...@gmail.com wrote:
Chad,
You are absolutely right on this one. I had setup the Queue time out for
agent set to 15 seconds and retry to 2 seconds. So, I think during those two
seconds Asterisk for some crazy reason hits another extension and then
On Fri, Nov 5, 2010 at 10:38 AM, Bruce B bruceb...@gmail.com wrote:
Yeah, I think I had it set to 2 seconds and that creates that short ring on
another extension.
Thanks,
The point was that 14 and 16 are divisible by 2 (evenly) while 15 is not.
--
On Fri, Nov 5, 2010 at 11:04 AM, Danny Nicholas da...@debsinc.com wrote:
Hi Gang,
My production box with my DAHDI cards is a 1.4.26 build. I
have 3 test machines that I do IAX communication with.
Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30.
Machine 2
On Sun, Oct 31, 2010 at 11:26 AM, Joel Maslak jmas...@antelope.net wrote:
If these are mobile users, I hope they never use any public networks
(hotels, starbucks) where other subscribers can do things like ARP attacks
to do MITM (and steal your calls; it might not be happening today, but it
On Tue, Oct 26, 2010 at 12:06 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
On 10/26/2010 05:52 PM, bakko wrote:
Hello,
many SIP phones offer you the possibility to provisioning them over a FTP
connection (with username and password).
Regards
- Bakko
In this case I will want to use
On Fri, Oct 22, 2010 at 10:02 AM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
For some reason a few phones connected to a pfSense box can't make or allow
in OpenVPN in port 1194 UDP. So, I established the VPN tunnel on 1194 TCP
and it works fine. I would like to know if there is any
I took a look in the source -- it is definitely asterisk -r (or
rasterisk) and not AMI. AMI logs something like Manager 'username'
logged on from 127.0.0.1.
Check the timing between calls and see if a pattern appears. If so,
it is some sort of cron/scheduled job. If not, keep looking!
-M
--
On Thu, Oct 14, 2010 at 11:46 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Crazy. What do you plan on using for an ATA now?
The problems I'm having are getting 500 Server Internal Error on just
about every other call placed out of this mp-118. The box has been
installed and in use
On Sat, Oct 16, 2010 at 5:36 PM, Dan Journo
d...@keshercommunications.com wrote:
Hi,
Does anyone know where this is suddenly coming from?
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
On Fri, Oct 15, 2010 at 11:02 AM, Danny Nicholas da...@debsinc.com wrote:
The original one is super quiet - obviously not Allison in a studio...
Listen to the gsm in Asterisk to see my quandary...
What is the end use here? Who will be listening to the recordings?
Users on PSTN and mobile
On Fri, Oct 15, 2010 at 11:20 AM, Danny Nicholas da...@debsinc.com wrote:
End use is Telephone Banking, so you've nailed the target audience.
BTW, the highpass and lowpass filters seem to help, but since I stopped
math at pre-calculus, the explanation of the Butterworth filter is beyond
my
2010/10/15 Matt Darnell mattdarn...@gmail.com:
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 10-10-15 04:10 AM, Сикорский Сергей wrote:
15.10.2010 9:40, Warren Selby пишет:
I think this means you need to set a call-limit for each sip peer
Is there any
On Tue, Oct 12, 2010 at 4:23 PM, Danny Nicholas da...@debsinc.com wrote:
dollars.gsm: data
dollars.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz
dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz
Can't be 100% certain on
Are you monitoring some dahdi hardware or a separate black box?
If dahdi, you could write a nagios plugin in shell with something like this:
ALARMS=`dahdi_scan | grep alarms | grep -v OK | wc -l`
and then set the appropriate exit code if ALARMS is not 0.
-M
On Tue, Sep 28, 2010 at 9:22 AM,
On Fri, Sep 17, 2010 at 11:52 PM, Dean Collins d...@cognation.net wrote:
Any thoughts on why the lack of traffic?
Cheers,
Dean
Not enough applications to play immature bathroom sounds.
Just a guess.
-M
--
_
-- Bandwidth
.
Best Regards,
Mark Deneen
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users
On Fri, Sep 17, 2010 at 11:51 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
On 09/17/2010 05:29 PM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 4:21 AM, Jonas Kellensjonas.kell...@telenet.be
wrote:
warning: exec file is newer than core file.
Jonas,
I encourage you to read the output
On Sun, Sep 12, 2010 at 8:57 AM, colin mcdermott
colinjamesmcderm...@gmail.com wrote:
Use lowercase for ftp:// . That might be the issue but it should be
easy to test. Do your FTP server logs shpw anything?
I will double check but I believe that lower case ftp is being used. I do
have
On Thu, Jul 1, 2010 at 12:53 PM, Tilghman Lesher tles...@digium.com wrote:
That would only be true if you used random characters in your 17-character
passphrase. In fact, English text has somewhere between 0.6 and 1.5 bits
of
randomness per letter, whereas an SHA1sum has no more than 4 bits
On Thu, Jul 1, 2010 at 7:09 PM, Miguel Molina mmol...@millenium.com.cowrote:
I've experienced a similar DTMF issue with recent asterisk 1.4 versions
(1.4.32, 1.4.33.1), I'm not sure about 1.6.2.X. What happens here is
that the DMTF activated features, like disconnect (default *) or blind
I do not believe the firewall is an issue.
Our ITSP is registered as a SIP provider, and we can receive calls just
fine. I've attached a file containing portions of the asterisk log, the
wireshark log and the dialplan.
Has anyone else run into this situation?
Best Regards,
Mark Deneen
[Jun 29
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