Hello,
Is there a way to trace into a log file, incoming AMI requests ?
For instance, I've got several apps accross the LAN, sending AMI requests
such as :
Action: originate
Channel: Local/7...@internal
Exten: 00123456789
Priority: 1
...
Some of them might be sometimes producing some erroneous
Hello list,
when the conversation is using the G729-codec and the conversation is
recorded with the Monitor()-application in wav-format, will there be
transcoding (and thus a need for licenses ?)
Kind regards,
Jonas.
--
_
Many of you are interested in and have used or recommended fail2ban
for your linux boxes. I finally installed it on our FreeBSD server (no
asterisk, hence the OT) with the help of a friend from the VoIP Users
Conference and Asterisk community.
After a lot of new learning about regex, I extended
On 13 July 2010 09:52, Randy R randulo2...@gmail.com wrote:
Many of you are interested in and have used or recommended fail2ban
for your linux boxes. I finally installed it on our FreeBSD server (no
asterisk, hence the OT) with the help of a friend from the VoIP Users
Conference and Asterisk
On Tue, Jul 13, 2010 at 11:04 AM, dotnetdub dotnet...@gmail.com wrote:
Hi Randy,
How many users are on this 'domain'? Google Apps Free is a great solution
for upto 50 users with 7.6GB per user. Their spam filtering usually does the
job for our customers.
Hi Brian,
Thanks for the reply. I'm
What I do, is only open port 25 to the list of ips of the spam filtering
service -- I use an iptables script called rc.firewall which I found
several years ago which works well and has a nice syntax for this and I
get no direct spam, I get some which gets by the filters.
Randy R
On Tue, Jul 13, 2010 at 12:29 PM, cov...@ccs.covici.com wrote:
What I do, is only open port 25 to the list of ips of the spam filtering
service -- I use an iptables script called rc.firewall which I found
several years ago which works well and has a nice syntax for this and I
get no direct
What you can do -- I don't know about nomad, but can you make them use
authentication?
Randy R randulo2...@gmail.com wrote:
On Tue, Jul 13, 2010 at 12:29 PM, cov...@ccs.covici.com wrote:
What I do, is only open port 25 to the list of ips of the spam filtering
service -- I use an iptables
On Tuesday 13 Jul 2010, Randy R wrote:
I was thinking of closing port 25 and using an alternate port (587?)
setup if the spam service is able to connect to an alternate port.
That way, the users can also change their configs to 587 and most
spammers will be trying 25 which is closed.
Can't
On Tue, Jul 13, 2010 at 12:53 PM, cov...@ccs.covici.com wrote:
What you can do -- I don't know about nomad, but can you make them use
authentication?
They do identify, but they have to connect first :)
--
_
-- Bandwidth and
On Tue, Jul 13, 2010 at 12:58 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Tuesday 13 Jul 2010, Randy R wrote:
I was thinking of closing port 25 and using an alternate port (587?)
setup if the spam service is able to connect to an alternate port.
That way, the users can also change
On Tue, Jul 13, 2010 at 4:29 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
when the conversation is using the G729-codec and the conversation is
recorded with the Monitor()-application in wav-format, will there be
transcoding (and thus a need for licenses ?)
I believe so, Yes. You can
Hi List,
I'm new to asterisk and currently running the newest of version. I'm
encountering the error below when I dial my meetme conference #:
WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo device
I already tried googling this issue and found some procedure but still no
luck
I have no licenses and I want to avoid transcoding all together.
When the phone supports G729 and the SIP provider support G729, then the
audio can just pass through...
However, in some cases the audio is recorded. Any change that we can
record in G729 format then ??
And how about
On Tue, 13 Jul 2010, Randy R wrote:
On Tue, Jul 13, 2010 at 12:58 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Tuesday 13 Jul 2010, Randy R wrote:
I was thinking of closing port 25 and using an alternate port (587?)
setup if the spam service is able to connect to an alternate port.
Marta Silva wrote:
Hi there,
Thank you for your response. So I can use the ModemSetOriginCmd
command to assign the outbound number on the iaxmodem, but how do I
choose which modem to use for my specific sip client (GXW-4004), as I
have 2 faxes connected to my GXW box?
Why would you need
Hi Gordon,
On Tue, Jul 13, 2010 at 1:55 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Technically/pedantically, users ought to be connecting to port 587 to submit
their email anyway, with port 25 being reserved for MTA to MTA
communications, so block 25 for everyone but the MX
Re-sent copying UNON and Expand Technologies. Apologies for the omission.
Rgds,
Alphonse
On Tue, Jul 13, 2010 at 3:27 PM, Alphonse Ogulla aogu...@gmail.com wrote:
Dear Esther,
The foregoing mail notes sent to Expand Technology refer, in view that you
were not copied in the initial
On Tue, 13 Jul 2010, Randy R wrote:
Hi Gordon,
On Tue, Jul 13, 2010 at 1:55 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Technically/pedantically, users ought to be connecting to port 587 to submit
their email anyway, with port 25 being reserved for MTA to MTA
communications, so
Did you mean to send this to a mailing list?..
S
On 13 Jul 2010, at 13:33, Alphonse Ogulla wrote:
Re-sent copying UNON and Expand Technologies. Apologies for the omission.
Rgds,
Alphonse
On Tue, Jul 13, 2010 at 3:27 PM, Alphonse Ogulla aogu...@gmail.com wrote:
Dear Esther,
The
On Tuesday 13 July 2010 06:35:45 Malvin Rito wrote:
Hi List,
I'm new to asterisk and currently running the newest of version. I'm
encountering the error below when I dial my meetme conference #:
WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo
device
I already tried
My sincere apologies for inadvertently sending this mail note to this list
which happens to be the first entry in my address book.
If you are a list administrator, kindly delete this thread from the list.
My apologies once again and please do not reply.
Rgds,
Alphonse
On Tue, Jul 13, 2010 at
On Tue, Jul 13, 2010 at 2:45 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Good luck!
A few have written me off list (thanks) so I thought I'd close out my
own thoughts on this. It's been about two hours and it does look
like things are working great. I removed the huge number of
On 07/13/2010 08:27 AM, Alphonse Ogulla wrote:
My sincere apologies for inadvertently sending this mail note to this
list which happens to be the first entry in my address book.
If you are a list administrator, kindly delete this thread from the list.
My apologies once again and please do
Hi Everyone,
I have done yum install speex libspeex-devel speex-devel and it was
succesful on CentOS. I then tried yum install asterisk16 asterisk16-addons
asterisk16-configs but core show translation doesn't show speex loaded.
Is there a way to or an option that I can append to the asterisk
Dan Austin dan_aus...@phoenix.com wrote:
Manmohan wrote:
My Web-MeetMe_v4.0.1, i followed the instructions in the
README File in the same package.
Good. There are other instruction packages, but since I wrote
the README it is the one I am most familiar with.
Are you using RealTime
)} - ${STRFTIME(${EPOCH},,%m)} -
${STRFTIME(${EPOCH},,%d)})
evaluates with what I expect:
-- Executing [7...@phones:4] Verbose(SIP/2625-d5f0,
20100713-110853 - 2010 - 07 - 13) in new stack
20100713-110853 - 2010 - 07 - 13
Is what I'm trying to do possible? It seems like it's at least recognizing
cov...@ccs.covici.com wrote:
Dan Austin dan_aus...@phoenix.com wrote:
Manmohan wrote:
My Web-MeetMe_v4.0.1, i followed the instructions in the
README File in the same package.
Good. There are other instruction packages, but since I wrote
the README it is the one I am most
},,%Y)} - ${STRFTIME(${EPOCH},,%m)} -
${STRFTIME(${EPOCH},,%d)})
evaluates with what I expect:
-- Executing [7...@phones:4] Verbose(SIP/2625-d5f0,
20100713-110853 - 2010 - 07 - 13) in new stack
20100713-110853 - 2010 - 07 - 13
Is what I'm trying to do possible? It seems like it's at least
On Tue, Jul 13, 2010 at 11:47 AM, Danny Nicholas da...@debsinc.com wrote:
You don’t state which version you are on (These things change from 1.2 to
1.4 to 1.6/8), but that being said, you would probably more likely to
succeed doing Set(GLOBAL) in an isolated context instead of using the
= ,n,Verbose(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} -
${STRFTIME(${EPOCH},,%Y)} - ${STRFTIME(${EPOCH},,%m)} -
${STRFTIME(${EPOCH},,%d)})
evaluates with what I expect:
-- Executing [7...@phones:4] Verbose(SIP/2625-d5f0,
20100713-110853 - 2010 - 07 - 13) in new stack
20100713-110853
2010/7/11 Julian Lyndon-Smith aster...@dotr.com
Anyone got a clue ? (he asks in desperation!)
Julian
On 9 July 2010 17:48, Julian Lyndon-Smith aster...@dotr.com wrote:
We are having a situation on our dialler here where our agents are
claiming that when they receive a call because it
On Tue, Jul 13, 2010 at 01:07:34PM -0400, Barry Miller wrote:
Try adding preload = func_strings.so to modules.conf
Ah, sorry. I just saw your earlier response that said you're on 1.4 -
I was remembering that after I migrated from 1.4 - 1.6, I had to preload
func_db.so so that I could use the
All:
Starting switching over my phone lines.
I got phone line 1 switched. Everyone working.
I switched the second phone line, and it worked about an hour, then I
started getting errors from the cli saying the server could not register
with the providing. I restarted the system, and it
On Tuesday 13 July 2010 11:30:44 Warren Selby wrote:
I'm trying to declare a few date-related global variables to ease my
dialplan. When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH =
On Tue, Jul 13, 2010 at 1:15 PM, Tilghman Lesher tles...@digium.com wrote:
When you load the dialplan, do you see the global variables getting set?
That would at least tell you whether the problem lies at the point where
the
values are loaded into memory, or later, at evaluation time.
==
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Tuesday, July 13, 2010 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] STRFTIME function declared in
On Tue, Jul 13, 2010 at 1:49 PM, Danny Nicholas da...@debsinc.com wrote:
--
Since you never know when you’ll need this, I slapped the code into my
1.4.30.
Here is the “corrected” code that works
YEAR = ${STRFTIME(${EPOCH}||%Y)}
MONTH = ${STRFTIME(${EPOCH}||%m)}
DAY =
On Tue, 2010-07-13 at 06:53 -0400, cov...@ccs.covici.com wrote:
What you can do -- I don't know about nomad, but can you make them use
authentication?
Randy R randulo2...@gmail.com wrote:
On Tue, Jul 13, 2010 at 12:29 PM, cov...@ccs.covici.com wrote:
What I do, is only open port 25 to
On Tue, 13 Jul 2010, Warren Selby wrote:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
On Tue, 13 Jul 2010, Danny Nicholas wrote:
YEAR = ${STRFTIME(${EPOCH}||%Y)}
Good catch. Looks like a bug to me.
Not that anybody cares, but the 2 statements exhibit the same bug in 1.2.
Just out of curiosity, why is
so nobody seems to like dealing with fax!!
2010/7/12 khalid touati khalidtou...@gmail.com
Hi Guys,
i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1)
and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue
i'm having is that i'm able to receive faxes
- khalid touati khalidtou...@gmail.com wrote:
so nobody seems to like dealing with fax!!
'Fax for Asterisk' is a commercial application sold by Digium. This is not
their official support channel. Since you paid for the product, why not contact
them directly about your problem?
On Tue, Jul 13, 2010 at 2:36 PM, Steve Edwards asterisk@sedwards.comwrote:
Good catch. Looks like a bug to me.
I'll create an issue on the tracker later today.
Just out of curiosity, why is the time the dialplan is reloaded of
interest?
Hi!
'Fax for Asterisk' is a commercial application sold by Digium. This is
not their official support channel. Since you paid for the product, why
not contact them directly about your problem?
Maybe because having to deal with Digium support is an ... uncomfortable
experience that I've made
On Tue, Jul 13, 2010 at 4:43 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
'Fax for Asterisk' is a commercial application sold by Digium. This is
not their official support channel. Since you paid for the product, why
not contact them directly about your
On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
I have no licenses and I want to avoid transcoding all together.
For terminating a call into Asterisk, you need g729 licenses. It is
that simple.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
I agree with horns you'll usually get better coverage. I have done
this in the past with 5 speakers for a 30k sq ft warehouse very good
coverage. Using bogen horns. This was for a 300ft by 100ft warehouse.
Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft
side I installed a horn
It has nothing to do with the D-channel, however you will never know
if the B-channels work if the D-channel is down. D-channel is what
allows the B-channels to work, and is the first place to troubleshoot.
If something is screwed up with the power the symptom you'll get is a
non working PRI, the
Thanks for the reply. There is no folder dahdi under /dev folder. I cannot
also find /udev.d on /etc folder.
Under /dev folder I only see /dev/zap/pseudo.
Regards,
Malvin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
hello
I want to know how to pass through 100rel header.
and I hope that asterisk PRACK to UAS.(RFC3262 behavior)
_
--
'Fax for Asterisk' is a commercial application sold by Digium. This is not
their official support channel. Since you paid for the product, why not
contact them directly about your problem?
i did get this version for free after buying a (actually several) digium
telephony card, but i realized that
I have a virtual server with godaddy but can not compile DAHDI as it
complains that I do not have the correct kernel source.
The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest
version
Nothing to do
uname -a
Thanks for the input guys. For other refrence, a CyberData Voip Amplifier
which supplies 10 Watt to each of the two bogen 30 Watt speakers did the job
for a 35, square feet warehouse with environmental noise level of
slightly higher than standard but not those of industrial.
Only two speakers
hello
I found silence RTP packet from Asterisk in early dialog.
I want to know reason and how to solve.
RTP packet
80 00 40 22 00 0c 74 58 06 98 eb 44 ff ff ff ff ..@..tX...D
0010 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
0020 ff ff ff ff ff ff ff ff
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