-users-boun...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
*Sent:* Friday, April 4, 2014 11:15 AM
*To:* Asterisk Users List
*Subject:* Re: [asterisk-users] Asterisk 1.6
Hello Ishfaq, outside users usually travel around the country and
connect from different network, so
real users
are located.
-=Michelle=-
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
Sent: Friday, April 4, 2014 11:15 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk 1.6
Hello
Hello All, my asterisk server is constantly under attack
[Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132
194.100.46.132:56714' - Wrong password
[Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673
I don't know what platform you are on, but if you are on Linux (and
possibly BSD) you could use fail2ban to block them at the network
interface.
On 04/04/2014 09:00 AM, motty cruz wrote:
Hello All, my asterisk server is constantly under attack
[Apr 4 06:56:00] NOTICE[21745]:
On 4 April 2014 15:00, motty cruz motty.c...@gmail.com wrote:
Hello All, my asterisk server is constantly under attack
Unfortunately you are not alone.
[Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register:
Registration from '4941 sip:4941@public_ip' failed for
On Fri, Apr 4, 2014 at 10:05 AM, Daniel Taylor dtay...@vocalabs.com wrote:
I don't know what platform you are on, but if you are on Linux (and
possibly BSD) you could use fail2ban to block them at the network
interface.
I second fail2ban. If you need some ideas to configure it, you
thank you all for your support. I am using Linux, I only have about 7 users
outside our home network. I will learn fail2ban and will use it
accordingly.
again Thanks for your support.
On Fri, Apr 4, 2014 at 7:09 AM, Mauricio Tavares raubvo...@gmail.comwrote:
On Fri, Apr 4, 2014 at 10:05
...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
Sent: Friday, April 4, 2014 10:00 AM
To: Asterisk Users List
Subject: [asterisk-users] Asterisk 1.6
Hello All, my asterisk server is constantly under attack
[Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register
On Friday 04 Apr 2014, Michelle Dupuis wrote:
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/
It does everything fail2ban does and more, including blocking users by
geography (we exclude all of Asia and Africa), detection of break-in
patterns (even if someone guessed
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk 1.6
On Friday 04 Apr 2014, Michelle Dupuis wrote:
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/
It does everything fail2ban does and more, including blocking users by
geography (we exclude all of Asia and Africa
absolutely right A J, thanks for the heads up.
I do not intent to implement that solution in production server, I hope to
learn it first, build a test server and monitor for a few days or weeks.
Thanks again,
On Fri, Apr 4, 2014 at 7:38 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On
On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote:
thank you all for your support. I am using Linux, I only have about 7
users outside our home network. I will learn fail2ban and will use it
accordingly.
again Thanks for your support.
Do the 7 users outside of your home
Hello Ishfaq, outside users usually travel around the country and connect
from different network, so it won't be possible to lock it down to specific
IP.
Thanks for your support.
On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On 4 April 2014 15:22, motty cruz
Well in that case fail2ban gets my vote.
On 4 April 2014 16:15, motty cruz motty.c...@gmail.com wrote:
Hello Ishfaq, outside users usually travel around the country and connect
from different network, so it won't be possible to lock it down to specific
IP.
Thanks for your support.
On
: Friday, April 4, 2014 11:15 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk 1.6
Hello Ishfaq, outside users usually travel around the country and connect from
different network, so it won't be possible to lock it down to specific IP.
Thanks for your support.
On Fri, Apr 4
Use allowguest=no
And define ACLs for every SIP account.
And obviously, fail2ban for blocking suspicious IPs.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
:* asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
*Sent:* Friday, April 4, 2014 11:15 AM
*To:* Asterisk Users List
*Subject:* Re: [asterisk-users] Asterisk 1.6
Hello Ishfaq, outside users usually travel around
-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Friday, April 04, 2014 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6
that sounds feasible, Thanks Michelle,
On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup
Hi All,
Currently i'm facing with a cdr issue, When i originate a call (outbound
call) to uncorrect/unregistration user, asterisk inform me that call was
failed but in mysl-cdr (cdr-csv also) records.
Here are 2 samples
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 Bad Extension back from 10.4.0.10
-- Stopped music on hold on
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Le 7/03/13 11:21, Steven Howes a écrit :
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
Hello Steve,
After checking, I confirm that the call is cut precisely to 900 seconds
(15 min).
10.4.0.1 =
Can u debug on AS ?
On Thu, Mar 7, 2013 at 9:20 AM, Mickael Monsieur
mickael.monsi...@gmail.com wrote:
Le 7/03/13 11:21, Steven Howes a écrit :
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
Hello
Le 7/03/13 11:12, Mickael Monsieur a écrit :
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 Bad Extension back
Hi,
voicemail plays after hitting # as final file auth-thankyou. Is
there any possibility to change this behaviour? Custom soundfile or
disable it perhaps?
Thanks for your answer(s)!
-Thorsten-
--
_
-- Bandwidth and
Update:
No luck with versions 1.6 and 1.8.7 I had to revert back to 1.4 which
worked with no problem.
Probably if I have some time, I will do more testing with version 1.8.7 to
see what the difference is and what changes need to be made for this kind
of setup to work in 1.8.7
Joseph
On Mon,
Hello,
Using asterisk 1.6 as sip client to register with sip provider and
terminate calls through them. SIP Provider has provided sip proxy and sip
server details. The problem is that the sip server FQDN does not resolve
on the internet. So I can only presume that the SIP proxy knows how to
Hi,
I have recently run into the problem with macro implementation in AEL in
Asterisk 1.6. I have some older AEL dialplan which runs on 1.4 but it does not
on 1.6 and I'm not sure how to solve this correctly. Let me explain...
For example, in Asterisk 1.4 I have a macro like this:
Hello everyone,
I am sure that someone can help with this. We decided to do a fresh install
of asterisk 1.6.2.19
And after we did that, the problem that we have is this - We cant run a
single Php file!
Here's the output:
-- Executing [8212@from-pstn:1] Answer(DAHDI/23-1, ) in new
On Sat, 16 Jul 2011, Zarko Zivanovic wrote:
I am sure that someone can help with this. We decided to do a fresh
install of asterisk 1.6.2.19 And after we did that, the problem that we
have is this – We cant run a single Php file!
testera.agi: Failed to execute
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi/system.conf
fxsks=1
# global data
loadzone = us
defaultzone = us
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi/system.conf
fxsks=1
# global data
loadzone = us
defaultzone = us
A couple things -
First, in extensions.con your context is [my-phone], but you're using my-phones
in your dahdi and sip.conf files.
Second, you need an 's' extension somewhere in your receiving context in order
for asterisk to answer the incoming analog call.
Third, I think you've got some
Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi
: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2
A couple things -
First, in extensions.con your context is [my-phone], but you're using
my-phones in your dahdi and sip.conf files.
Second, you need an 's' extension somewhere in your receiving context in
order for asterisk to answer the incoming
Hi all,
I try to figure out why I have empty :
sip show subscriptions
list in may asterisk 1.6.
When device is registering to asterisk I can see in log:
NOTICE[25603]: chan_sip.c:21518 handle_request_subscribe: Received SIP
subscribe for peer without mailbox: 1010
but
sip show subscriptions
http://pastebin.com/vxGM2n5j
We are getting those errors 100x per second in console when AGI set debug is
on
It is causing extremely high CPU usage, we've tried asterisk version
1.6.1.22 and 1.6.2.18
It seems the problem is worse in 1.6.2.18
Can someone advise how to fix this? Thank you.
--
On Sun, 5 Jun 2011, vip killa wrote:
http://pastebin.com/vxGM2n5j
We are getting those errors 100x per second in console when AGI set
debug is on
Can someone advise how to fix this? Thank you.
Don't request 'WAIT FOR DIGIT 1000' from a dead channel.
Don't ignore the error from 'WAIT
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom
List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6: Custom Name for
Recordings file
Hi,
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
RSCL Mumbai
Sent: Friday, May 13, 2011 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6
On 5/13/11 10:57 AM, RSCL Mumbai wrote:
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like
I have a couple of questions about asterisk 1.6:
Can you change codecs mid-call upon re-invite?
Can you handle the SDP offer-answer in the 200-ACK instead of the usual
INVITE-200?
Thanks in advance for any insight.
Gary
--
On 05/03/2011 12:43 PM, Gary Graves wrote:
Can you change codecs mid-call upon re-invite?
Do you mean: can Asterisk be configured to _initiate_ such a change
at some point, mid-call? Or do you mean: Will Asterisk properly
react to such a re-INVITE and change codecs if asked to do so by
Can you answer both?
Can Asterisk be configured to _initiate_ such a change at some point,
mid-call?
and
Will Asterisk properly react to such a re-INVITE and change codecs if asked
to do so by the dialog counterparty?
On Tue, May 3, 2011 at 12:56 PM, Alex Balashov
On 05/03/2011 01:16 PM, Gary Graves wrote:
Can you answer both?
Can Asterisk be configured to _initiate_ such a change at some point,
mid-call?
I don't know of a way to do that. I suppose it might be possible if a
call were asynchronously transferred to a SIP peer that had different
codec
Hi
very thanks, that's work
bye
olivier
2011/4/3 Mark Murawski markm-li...@intellasoft.net:
I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten = s,1,Playback(silence/1)
On 04/03/11 14:14,
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten =
In that situation, I've had to do a pickup macro that kind of primes
the audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the
callee's channel
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3
I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten = s,1,Playback(silence/1)
On 04/03/11 14:14, Olivier CALVANO wrote:
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
Okay, so here's the configuration I have for MySQL Realtime (Asterisk
version 1.6.2.17):
In /etc/asterisk/extconfig.conf:
sipusers = mysql,mya2billing,cc_sip_buddies
In /etc/asterisk/res_mysql.conf:
[mya2billing]
dbhost = localhost
dbname = mya2billing
dbuser = a2billinguser
dbpass = REDACTED
On Mon, Mar 7, 2011 at 5:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
Okay, so here's the configuration I have for MySQL Realtime (Asterisk
version 1.6.2.17):
In /etc/asterisk/extconfig.conf:
sipusers = mysql,mya2billing,cc_sip_buddies
In /etc/asterisk/res_mysql.conf:
Don't know
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
[mya2billing]
dbhost = localhost
dbname = mya2billing
dbuser = a2billinguser
dbpass = REDACTED
dbport = 3306
Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config
stanza and see if that helps (or
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca
wrote:
[mya2billing]
dbhost = localhost
dbname = mya2billing
dbuser = a2billinguser
dbpass = REDACTED
dbport = 3306
Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config
stanza and see if that helps
Hello folks,
for a customer of us we are trying to make asterisk and windows RTC
library work together, but without success.
We *need* to use gsm codec, so in the peer section we have
disallow=all
allow=gsm
the sip signaling is ok, and when sniffing we got this session description:
INVITE from
2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote:
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so
On Monday 24 January 2011 04:09:31 Olivier wrote:
2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote:
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script
On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote:
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so when asterisk is
running and I ssh tot he server then
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so when asterisk is
running and I ssh tot he server then 'asterisk -vr' to attach to the
asterisk console there are no colors. If I use the safe_asterisk
script to start
On 20 January 2011 18:01, JR Richardson jmr.richard...@gmail.com wrote:
Or is there another work around to get ssh console colors using the
Debian * 1.6.0.28 LSB init script?
I also tried 'nocolor = no' in the [options] section of asterisk.conf
with no effect.
Try running asterisk using
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so when asterisk is
running and I ssh tot he server then 'asterisk -vr' to attach to the
asterisk console there are no
Hi,
I had 2 Asterisk servers connected together in iax with auth=rsa and
proper keys for user and peer in each direction. It worked well till I
upgraded one of them to Asterisk 1.6.13 Since I get No authority found
I thought that problem came from keys as the server with 1.6.13 was
changed
On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote:
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either the Asterisk server
On Wednesday 22 December 2010 08:23:19 MrHanMan wrote:
On Wed, Dec 22, 2010 at 6:41 AM, Steve Davies davies...@gmail.com
wrote:
On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es
wrote:
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk
Am 20.12.2010 21:39, schrieb Ernie Dunbar:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death), or
the kernel runs out of PID
Am 20.12.2010 21:39, schrieb Ernie Dunbar:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either
the Asterisk server is restarted (and the zombies die a natural death),
or
the kernel runs out of
Your server is being brute-forced. Read this article (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) and set up fail2ban on your machine right now.Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS -
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death),
Actually, no. This is part of a migration, and those are mostly customers'
secondary lines (which for the most part, aren't even active). We get a
lot of these bad logins because the retry times on the ATAs are quite
short.
Asterisk really *shouldn't* leave zombies around for every bad login, but
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either the Asterisk server is restarted (and the zombies die a natural
death), or the kernel runs
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca
wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either
the Asterisk server is restarted (and the zombies die a natural
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death), or
the kernel runs out of PID space (happens within hours) and brings the
Hi,
I am really not sure if this is related to the meetme in asterisk OR
this is something to do in web-meetme. I tried to google but didnt get
any proper results.
I am facing one issue in Web-meetme on the expiry of any conference
that we create.
I am having Web-meetme 4.0.2 over Asterisk 1.6
Manmohan wrote:
I am really not sure if this is related to the meetme in asterisk OR
this is something to do in web-meetme. I tried to google but didnt get
any proper results.
I am facing one issue in Web-meetme on the expiry of any conference
that we create.
I am having Web-meetme 4.0.2
Hi Dan,
I am currently on Asterisk 1.6.2.14.
Thanks Regards
Manmohan Singh
On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin dan_aus...@phoenix.com wrote:
Manmohan wrote:
I am really not sure if this is related to the meetme in asterisk OR
this is something to do in web-meetme. I tried to google
Manmohan wrote:
I am currently on Asterisk 1.6.2.14.
Do you have schedule=yes in meetme.conf? I incorrectly
remembered/thought that all of the Realtime features were
controlled by that option, only a small number, such as
end times and CDR logging
On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin
Hi Dan,
In meetme.conf the schedule=yes was commented, after removing its working fine.
But one strange thing had started now. I started getting segmentation fault.
following are the errors which i see in it:
warning: difference appears to be caused by prelink, adjusting expectations
: [asterisk-users] Asterisk 1.6 (Web-meetme)
Hi Dan,
In meetme.conf the schedule=yes was commented, after removing its working fine.
But one strange thing had started now. I started getting segmentation fault.
following are the errors which i see in it:
warning: difference appears to be caused
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
--
No you can't
On Wed, Nov 24, 2010 at 2:34 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
in extensions.conf:
exten = 0532xx,1,Answer
exten = 0532xx,2,MusicOnHold(Sound_1)
exten = 0532xx,3,Dial(SIP/ACCOUNT001,180,t)
exten =
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
in extensions.conf:
exten = 0532xx,1,Answer
exten =
2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com:
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
in
On Wed, Nov 24, 2010 at 5:35 AM, Olivier CALVANO o.calv...@gmail.com wrote:
2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com:
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
. Johansson o...@edvina.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 11:30:27 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
10 nov 2010 kl
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark
5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Set(SIP/413-0005, CALLERID(num)=2) in
new
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asterisk-users@lists.digium.com
Sent: Wednesday, November 10, 2010 8:47:38 AM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote: Good idea Paul.
My debug
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
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Subject: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num)
Problem
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum br...@woollum.com wrote:
Nobody has any idea why the Caller ID is being overwritten when using
Asterisk Realtime for the SIP users?
No, perhaps you can _show_ us the problem.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
- Original Message -
From: Paul Belanger paul.belan...@polybeacon.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 5:18:36 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime
10 nov 2010 kl. 02.38 skrev Brett Woollum:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9
Hello,
I got the time to upgrade my tutorial about Asterisk and Kamailio
realtime integration to latest stable release of Kamailio, version 3.1.0
(out on Oct 6, 2010).
You can find the document at:
*
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
Besides making
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point everything works in terms of phones registering, placing calls
between them, etc. However, I am having a problem
Has anyone started using Firefox 4 beta versions? We started today and find
that many of the GUI's attached to Asterisk respond differently and in many
cases not at all? We have found that details cannot be saves and that the
screens become very unstable. While we appreciate this is a beta
Hello,
I'm experimenting with Overlap Dialling in asterisk 1.6.
I've enabled this in sip.conf and on the SNOM 300 phone.
My problem is that asterisk dials out as soon as it matches an
extension without waiting to see if the user is going to type in more
digits.
Is there a way to set a timeout
Hello list,
are there special things that needs to be done when converting BLF from
asterisk 1.4 tot 1.6.2 ?!
I have replaced call-limit with call-counter, but it seems that the
lights on the phone no longer give the status of the extension they monitor.
On Snom phones, when the lights
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
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