Slawek Sloma wrote:
> Is there an option in asterisk to manipulate the audio in a call?
> I would like to, for example change the voice of one caller but
> without manipulating the audio that comes from another caller.
> I have read about something called JACK but i don't know if i can use it
> fo
On Apr 20, 2010, at 5:18 PM, Frank Bulk wrote:
> Please take note of their posting:
> https://aws.amazon.com/security/
> which discusses the issue and what they're doing to improve response.
This is an incredibly lame post on their part. They go out of their way to
point out there was n
I agree, our "quickly" and Amazon's "quickly" are two different things.
Maybe it was quickly for them. And note that they say "when *we* find
misuse". Even though a customer may have identified it, their AWS abuse
(team?) may not run a 24x7 operation and further delay things.
Frank
-Origina
On Tuesday 20 April 2010 20:22:39 Alejandro Recarey wrote:
> Hi all,
>
> I am having trouble getting cdr_adaptive_odbc to work.
>
> I have correctly configured the odbc drivers and dsn (I have tested
> this by connecting directly using isql). I have also configured
> /etc/asterisk/cdr_adaptive_odbc
Hi all,
I am having trouble getting cdr_adaptive_odbc to work.
I have correctly configured the odbc drivers and dsn (I have tested
this by connecting directly using isql). I have also configured
/etc/asterisk/cdr_adaptive_odbc.conf like so:
[test-asterisk]
connection=test-asterisk-odbc
table=cdr
Doug, thanks for the help, already looked it up, but it does not seem
to be a NAT issue (which is what most posters suggest when googling)
Danny, those are billsec durations, the call has been established and
media is being passed for 20 seconds.
Thanks again!
Alex
--
_
Carlos Chavez wrote:
> On Tue, 2010-04-20 at 15:04 -0400, Leif Madsen wrote:
>> Carlos Chavez wrote:
>>> I have a customer that needs to record all calls coming in and out.
>>> The problem I am having is when a call comes in to the operator and it
>>> is transferred to another extension. The f
On Apr 20, 2010, at 6:18 PM, Frank Bulk wrote:
> Please take note of their posting:
> https://aws.amazon.com/security/
> which discusses the issue and what they're doing to improve response.
>
> Frank
>
If only they wrote the truth...
"When we find misuse, we take action quickly and shut
Please take note of their posting:
https://aws.amazon.com/security/
which discusses the issue and what they're doing to improve response.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred Posne
Hello Asterisk Community,
Does somebody had used an IBM X3650 server with Asterisk? we would like to
know if this servers are reliable and works OK with linux and Asterisk?
Does some of you has this servier with asterisk on production? should we
install an special BIOS (linux bios) in order to ma
Hi All,
Is there an option in asterisk to manipulate the audio in a call?
I would like to, for example change the voice of one caller but
without manipulating the audio that comes from another caller.
I have read about something called JACK but i don't know if i can use it for
this (or how to use
On Tue, 2010-04-20 at 15:04 -0400, Leif Madsen wrote:
> Carlos Chavez wrote:
> > I have a customer that needs to record all calls coming in and out.
> > The problem I am having is when a call comes in to the operator and it
> > is transferred to another extension. The first mixmonitor begins
>
>
> Further to Steve Edward's comment, I think things would be more
> obvious if the help system was improved slightly, for instance:
>
> If you were trying to figure out the commands dealing with peers, you
> would be able to type:
> *CLI> help peer
> No "peer" command found. Possible alternati
Steve Edwards wrote:
> How obvious.
>
> Kind of makes me wish I still used 1.2 -- oh wait, I do.
If you're used to the 1.2 commands, look at cli_aliases.conf. It has templates
for 1.2 and 1.4 CLI commands.
It does seem we've missed the 'soft hangup' command though. I'll add that now.
> Serious
Carlos Chavez wrote:
> I have a customer that needs to record all calls coming in and out.
> The problem I am having is when a call comes in to the operator and it
> is transferred to another extension. The first mixmonitor begins
> recording when the operator picks up but the recording stop
When the
Portech MV-374 is connected directly to the internet, then there is no
problem. So I guess this is a NAT-problem as I started to suspect...
I'm using a Zyxel NGB-419 as VoIP-router, but I don't know what
configuration I exactly need to do now... I guess I need to define some
static ma
Hi,
This problem has been tackled over and over, I know.
I'm trying to understand why I'm having trouble with my "simple setup".
My setup is like this:
-
I've unloaded the nf_*_sip kernel modules from the LINUX_GATEWAY just in case
they could interfere.
The DSL1 modem/router is a THO
We have Asterisk 1.4.23.1 running on a Dell rack server and the audio is not
playing initially. Seems to be throughout, if you dial *65 we only hear the
last bit of the extension and the voicemails being left are missing the
beginning of the message. Running the same load of software we have run
On Tue, 20 Apr 2010, Jared Smith wrote:
> On Tue, 2010-04-20 at 09:49 -0700, Steve Edwards wrote:
>> I'd like to see a more natural and intuitive interface following a "verb
>> noun" model like Oracle, MySQL, or even GDB.
>
> We're close to that now, and that's one of the reasons that the "soft
>
On Tue, 2010-04-20 at 09:49 -0700, Steve Edwards wrote:
> I'd like to see a more natural and intuitive interface following a "verb
> noun" model like Oracle, MySQL, or even GDB.
We're close to that now, and that's one of the reasons that the "soft
hangup" command was changed to "channel request h
Thanks.
On 4/13/2010 3:07 AM, Ioan Indreias wrote:
> On Mon, Apr 12, 2010 at 8:19 PM, Mark Hulber
> wrote:
>
>> I want to use a voip speaker phone as a room monitor. Requirements:
>>
>> A phone that I can set to auto answer in speaker mode.
>> A phone with a good speaker phone.
>> Ability t
On Tue, Apr 20, 2010 at 10:49 AM, Steve Edwards
wrote:
> On Tue, 20 Apr 2010, Tilghman Lesher wrote:
>
>> On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote:
>>> I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI>
>>> prompt, and found references on using the command "soft ha
I have a customer that needs to record all calls coming in and out.
The problem I am having is when a call comes in to the operator and it
is transferred to another extension. The first mixmonitor begins
recording when the operator picks up but the recording stops when the
call is transfer
On Tuesday 20 April 2010 11:49:47 Steve Edwards wrote:
> On Tue, 20 Apr 2010, Tilghman Lesher wrote:
> > On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote:
> >> I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI>
> >> prompt, and found references on using the command "soft ha
On Tue, 20 Apr 2010, Tilghman Lesher wrote:
> On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote:
>> I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI>
>> prompt, and found references on using the command "soft hangup
>> ", but as you can see below, the "soft hangup" comma
On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote:
> I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI>
> prompt, and found references on using the command "soft hangup
> ", but as you can see below, the "soft hangup" command
> does not seem to exist, and there is no mention
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Thanks all for your input, much appreciated.
I will investigate this further with The Google :)
On 20/04/2010 16:27, Steve Edwards wrote:
> On Tue, 20 Apr 2010, Bruce McAlister wrote:
>
>> Incidently we are currently running 1.4.x for our voice ma
If you do not put a context in the beginning of the sip.conf file, the
default is, ta da, default in extensions.conf. Putting a context=testof
idea in sip.conf got things moving:
sip.conf
[general]
port=5060
bindaddr=0.0.0.0 ;10.8.0.34
*context=testofidea*
srvlookup=yes
disallow=all ;read som
Try changing port=5064 to port=5060 in your Asterisk config file. Portech
will negotiate it's port with Asterisk itself.
On Tue, Apr 20, 2010 at 10:50 AM, Jonas Kellens wrote:
> When there is a register with bad password, then this is the SIP response
> :
>
> <>
> [Apr 20 16:44:29]
>
Maybe "core soft hangup"?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Johnson
Sent: Tuesday, April 20, 2010 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
Hello asteriskers,
I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI>
prompt, and found references on using the command "soft hangup
", but as you can see below, the "soft hangup" command
does not seem to exist, and there is no mention about it in the
UPGRADE*.txt documents.
C
Hi
There is a setting in sip.conf that deafults to
context=default ; Default context for incoming calls
As explained this means that all incoming calls go to the default
context. You can change this if you wish to but just remember that
incoming calls will only go into one context and
On Tue, 20 Apr 2010, Bruce McAlister wrote:
> Incidently we are currently running 1.4.x for our voice mail server,
> however if 1.6.x offers maxsecs (and others) as a configurable, per
> mailbox, setting then we will look in to upgrading the environment.
>
> Can you, or any else, confirm that maxs
Looking at the 1.6.1.6 ChangeLog, I see that Tilghman Lester made changes to
this in the 1.6 branch in 2006. According to the docs, maxsecs should be
functional in all 1.6 branches after 12-31-2006.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-bo
Message: 15
Date: Mon, 19 Apr 2010 17:46:46 -0400
From: Ryan Bullock
Subject: Re: [asterisk-users] A matter of context
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID:
Content-Type: text/plain; charset="iso-8859-1"
Have you tried 'type = friend', might also
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Hi Steve,
Thats interesting, I had a look through versions 1.6.0, 1.6.1 and 1.6.2
and I didnt see anything mentioned that maxmessage (maxsecs) has been
enabled on a per mailbox setting. I did see that the maxmessage setting
was renamed to maxsecs so t
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Hash: SHA1
Hi Jared,
I'm talking about the maxmessage setting which is the maximum amount of
time that a voice message can be, not the maximum number of messages in
a mail box.
Thanks
Bruce
On 20/04/2010 15:51, Jared Smith wrote:
> On Tue, 2010-04-20 at 14:34
Hi,
What would you say about adding this feature ?
The idea, in a free sitting or call center environment, is to warn an agent
that the ongoing call will be timed out within a given time frame.
Options are :
the file name to play to warn the agent,
the duration before hanging the call.
Suggestio
On 04/19/2010 03:48 AM, Peter Gelencser wrote:
> I've run into a veird problem. I'm using a B400P BRI and an A1200P card
> with dahdi (2.2.1) driver. The dahdi_scan shows the each moduls and
> spans, everything seems fine. With dahdi_genconf I made the config, set
> up the channels in chan_dahdi
This should work:
exten => s,n,Gotoif($[${LEN(${ACCEPT})} < 1]?no)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Tuesday, April 20, 2010 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
When there is
a register with bad password, then this is the SIP response :
<>
[Apr 20 16:44:29]
<--- Transmitting (NAT) to my_public_ip:5061 --->
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP
192.168.1.22:5061;branch=z9hG4bKbdc15a66904d1239;received=my_public_ip
From: "test"
Hi Danny,
There seems to be a syntax error with your solution but i can't pin point it.
This is what happens when I hit "1".
-- Executing [...@macro-screen:2] Read("SIP/magrathea-1ec6",
"ACCEPT|priv-instruct-custom|1") in new stack
-- Accepting a maximum of 1 digits.
-- Playing
Hello,
Let say users A and B are on call.
>From dialplan, I would like to check if user A is on hold ?
With CHANNELS(), I can list ongoing channels.
With IMPORT(), I can read various channels variables but I can't any that
matches musiconhold status.
(I'm looking for something like IMPORT(SIP/111
On Tue, 2010-04-20 at 14:34 +0100, Bruce McAlister wrote:
> Is it at all possible to have the "maxmessage" setting on per
> user/mailbox value?
Absolutely, as long as you're talking about the "maxmsg" setting! In
fact, there's an example in the sample voicemail.conf file that comes
with Asterisk:
On Tue, 20 Apr 2010, Bruce McAlister wrote:
> Is it at all possible to have the "maxmessage" setting on per
> user/mailbox value?
I'm a 1.2 Luddite, so YMMV...
In 1.2, maxmessage is a "global" setting.
In 1.6.1.6 (just what I happened to have on hand), maxmessage has been
renamed to maxsecs a
On 04/19/2010 10:41 AM, Christian Hiller - Baig Tel LTD wrote:
> i am using a TC400B transcoding card, and sometimes when a G723 call is
> coming in, that is getting transcoded to G711, the CLI is flooded with
> ..
> [Apr 19 17:39:32] WARNING[3336] translate.c: zapg723toslin did not
> update samp
FYI, you could run a second, third, 4th, etc. asterisk on the same box with
the sole purpose of the "other" asterisks being to serve up the IAX
connection and custom voicemail capability. Or the "second" asterisk could
run an AGI to reload voicemail.conf for each caller, changing maxmessage on
an
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Hi Danny,
Thanks for the tip, although a second instance of asterisk, in our
environment, could very well mean additional hardware to offer this
feature for a select few users. In some cases users want variable
options, leaving 30 minute long VM's, so
just FYI, to complete the topic.
The problem was caused by failed PVDM module in Cisco server.
> Hello,
> I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to
providers Cisco 2800 with VWIC-1MFT-E1 card.
> the same card runs fine with another E1 provider.
> TE420 led
Bruce,
thank you for your answer. I have not changed the default login &
password of the MV-374...
In sip.conf, I have this for the SIM 3 :
[simsim3]
type=friend
host=dynamic
username=simsim3
secret=xxx
port=5064
insecure=port,invite
context=from_SIM
disallow=all
allow=alaw
allow=gsm
qua
The "Out of the box" answer is no. A "simple" workaround would be to have a
second instance of Asterisk that you connect to via IAX to let the "special"
group leave a longer message.
Exten => s,1,noop(voicemail processing)
Exten => s,n,Gotoif(..special..)?longmail
Exten => s,n,Voicemail(${ext...@
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Hi All,
Is it at all possible to have the "maxmessage" setting on per
user/mailbox value?
We have a requirement whereby we want the global maxmessage setting to
be 180 seconds per mail box, however, we would like to have certain
users to be able to s
I have had problems with Portech firmware using Chrome browser. The problem
was that when I changed the password on the gateway it would apply that
password to SIP PEERS as well. So, maybe, you are actually not having the
right password in your SIP peer as well and hence your Asterisk sends
Unautho
>From my exposure, this is a bridging issue; you have a Dial command with a
20 second timeout. If the call does not bridge or complete in 20 seconds,
Asterisk considers it completed as failed. If you change the Dial to 30
seconds, the problem will become a 30 second one, etc...
-Original Mes
With tcpdump
I saw that there were packets coming in from the GSM-gateway to the
public Asterisk-server.
I saw nothing on the Asterisk-CLI that told me that there were attempts
to register, but a "sip debug" shows this :
<>
[Apr 20 15:07:41] Scheduling destruction of SIP dialog
'0c
Do this:
exten => s,n,GotoIf($["${ACCEPT}" = "1" ] ?yes:no)
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 20, 2010, at 6:11 AM, Dan Journo wrote:
> Hello,
>
> I use the following macro to screen calls when they come in.
>
> Priv-instruct-custom says “press 1
Not necessarily the best way, but here's how I handle this type of thing in
1.4.X
macro-screen]
exten => s,1,Wait(0.2)
exten => s,n,Read(ACCEPT|priv-instruct-custom|1)
exten => s,n,GotoIf($[LEN(${ACCEPT}) < 1 ] ?no)
exten => s,n,GotoIf($[${ACCEPT} = 1 ] ?yes:no)
exten => s,n(no),Set(MACRO_R
Hello,
I use the following macro to screen calls when they come in.
Priv-instruct-custom says "press 1 to accept, press 2 to reject"
However, when no input is made (or the call goes to my mobile's voicemail and
therefore no input is made), the result is that the ACCEPT variable is not set
and
Just a WAG - the speaker button press in the on-hook is being interpreted as
a "flash", resulting in 2 dial actions.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Brady
Sent: Tuesday, April 20, 2010 3:57
On Thu, Apr 15, 2010 at 4:06 PM, Baji Panchumarti <
baji.panchuma...@gmail.com> wrote:
> Steve, Chris :
>
> I too had this problem and the solution was not tweaking
> the AMD parameters, but playing a short audio file (even
> a really really short one) before executing the AMD function.
>
> T
We use park to "hold" the call and then have an extension that we call to
retrieve the parked call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 19, 2010, at 11:48 PM, Thermal Wetland wrote:
> I would like to be able to place a call on hold via the manager inte
Hello list,
has anyone experience with the Portech MV-374 GSM-gateway ?
I'm trying to register the SIP-accounts to a public SIP-server but that
fails.
When trying to register to a local Asterisk-server, all goes well.
So anyone knows what special setting I need to make to register my
SIP-ac
Hi Robert,
The opensips dialog module mainly does dialog monitoring and has limited
capability when comes to checking dialog health (like it the call is not
zombie and it is really ongoing). The dialog module can just expire too
long calls (using a timeout for call duration).
First of all, de
(sorry this is so long)
I could really use a helping hand. I have a 1.6.2.6 installation using
LDAP as the realtime engine for voicemail users, SIP users, queues, and
some custom hotdesking families. I'm also using ODBC voicemail storage.
The issue that I am having is that the UA's (Polycom 5
On 04/19/2010 02:22 PM, Jay Vocaire wrote:
> I have searched everywhere, but cannot seem to find anyone else talking about
> this issue. Maybe I am just using the wrong search terms.
>
> I am running Asterisk 1.6.2 and multiple Polycom phones all with 3.2.3 (the
> latest) firmware on them.
>
>
Todays SIP attack blast comes to you from:
inetnum:89.255.0.0 - 89.255.63.255
netname:NL-HOSTING-CONCEPTS-20060914
descr: Netnation Europe V.O.F.
country:NL
Specifically in my case: 89.255.8.146
Gordon
--
___
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