Hello,
I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it
in wav-audio at the Asterisk server. I found the loudness level of the
recorded audio was too high comparing with the orginal audio. How can I
ajust it, so that there will be no amplifier used for recording.
Thanks
Am 03.03.2011 16:02, schrieb satish patel:
Hey Guy,
I have quick question. I am purchasing Sangoma A102D card but i am
confused between PCI and PCI Express. Which card would be good for
me.
Definitely PCI Express is advance but i
Danny - Thanks, but that wouldn't work either - as I am fetching
multiple rows (not in that example - but I do in a production
environment).
Steve - If mySQL in the dialplan is so bad - why did Digium include it
in the first place? JFYI - I use mySQL in the dialplan all the time -
and it always
On Thursday 03 March 2011 08:42:42 Andrew Thomas wrote:
Does anybody know of a way to test whether a mySQL connection invoked
from the dialplan is current or not?
There is no way to test it. If you want this, you should track the
information yourself or don't disconnect anywhere but in the h
On Friday 04 March 2011 02:47:56 Andrew Thomas wrote:
If mySQL in the dialplan is so bad - why did Digium include it
in the first place?
Digium is not responsible for everything that appears in Asterisk. This is
a community project, and community volunteers have written large swaths
of
On Thu, 2011-03-03 at 08:19 -0800, Steve Edwards wrote:
Try something 'simpler'
mpg123 -q -w ${TEMP} ${INPUT}
sox ${TEMP} -c 1 -s -w -r 8000 ${OUTPUT}
and see if that helps. Otherwise, how do the 'intermediate' files in
your
process sound? Can you hear when things fall
Thanks Tilghman - this is exactly what I wanted to hear. As for the
'inclusion' bit - true, but it's still infused in to the addons package
at the Digium end (isn't it?).
Anyway, I'll go create a mysql.conf file now :)
Cheers
-Original Message-
From:
On Friday 04 March 2011 03:03:41 Andrew Thomas wrote:
Thanks Tilghman - this is exactly what I wanted to hear. As for the
'inclusion' bit - true, but it's still infused in to the addons package
at the Digium end (isn't it?).
While Digium hosts the repository and the project head (Russell) is
On Thursday 03 Mar 2011, Piotr Górski wrote:
As free I mean no subscription. I can write AGI that will query
numberingplans.com - that's not a problem... but I can query site only 20
times a day without a subscription... So it's not free.
Well, free is as free does :)
For the time being,
Problem as follows:
[default]
exten = 777,1,Gosub(sub,1,1)
exten = 777,n,Hangup()
exten = h,1,NoOp(hung up in 'default' context)
[sub]
exten = 1,1,NoOp(in sub)
exten = 1,n,Playback(tt-monkeys)
exten = 1,n,Return()
exten = h,1,NoOp(hung up in 'sub' context)
This works fine if the caller listens
Nevermind - I've re-written my dialplan so that all subs are in one
context. Now I only need 1 more line of code.
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: 04 March 2011
Hey all,
Alright. So we decided to not go with Avaya for our next PBX and we are now
full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP
gateway and call center and Lync is our internal UC and IP-PBX server. I've
already got Asterisk tied with our Nortel/Merridian
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 2:31 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Loudness of recorded wav-audio
Hello,
I sent a wav-audio to
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro
Sent: Friday, March 04, 2011 8:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk - Lync / Call Center Transfer / Refer
Thank you! How can I reduce the RXgain?
Am 04.03.2011 um 15:21 schrieb Danny Nicholas da...@debsinc.com:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 2:31 AM
To:
In sip.conf, add rxgain=-4.0 to the peer. This (feel free to correct)
should reduce the incoming volume by 4 decibels. You'll have to do a sip
reload for this to take effect.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Hi,
I have an issue with a GWX4004 used a as a VoIP trunk to PSTN lines
converter. In some instances, lines get stuck (both parties hang up, but
the GXW4004 status shows off hook for the lines). It stays like this until
reboot.
Is there a specific setting I should be looking for? I
Could yoz tell me the default value of rxgain or txgain, if there is no rxgain
or txgain in conf-data defined?
Von meinem iPad gesendet
Am 04.03.2011 um 15:34 schrieb Danny Nicholas da...@debsinc.com:
In sip.conf, add rxgain=-4.0 to the peer. This (feel free to correct) should
reduce the
Defaults are 0.0 (leave volume unchanged) +values make volume louder, -
softer.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Friday, March 04, 2011 8:55 AM
To: Asterisk Users Mailing List -
PCI Express is always good also it take less power and faster when compare
to PCI in interupts .
On Fri, Mar 4, 2011 at 2:13 PM, Thorsten Göllner t...@ovm-group.com wrote:
Am 03.03.2011 16:02, schrieb satish patel:
Hey Guy,
I have quick question. I am purchasing Sangoma A102D card but i am
Danny Nicholas wrote:
In sip.conf, add rxgain=-4.0 to the peer.
The last I knew, rx/tx gains are only for dahdi/zaptel devices.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Friday, March 04, 2011 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Loudness of recorded
Piotr Górski pi...@prnet.pl writes:
So how to bill customers? Number portability makes it pretty impossible...
In the US, you pay the same to call a cell phone as you pay to call any
other phone. The callee pays for the airtime. This is a sensible
arrangement, as it allows for number
I feel your pain
On Fri, Mar 4, 2011 at 9:29 AM, 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 Louis
Carreiro
Sent: Friday, March 04, 2011 8:07 AM
To:
Hi every one,
I am doing some experiments on asterisk server
performance.. How can we know server performance? can any one explain me
plz
I have 2 doubts regarding the asterisk server performance...
1. When can we know asterisk server performance?
1. when server
On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna
viswavardhanre...@gmail.com wrote:
Hi every one,
I am doing some experiments on asterisk server
performance.. How can we know server performance? can any one explain me
plz
I have 2 doubts regarding the
HI,
The way you said is correct, we are using SIPp to generate as many
calls as it can send and and the server is able is to take simultaneously of
560 - 570 calls
1. when we kept server for some time as idle it took 575 calls
2. when we kept again server as busy by continous calls back
Hi,
I mean when the cpu history is in idel and in busy state...
i have one more doubt that we are doing experiments on server
performance(only on software) it does not depends on hardware or even on
systemm/...
knowing the server performance only the software side includes any
Ha! Thanks Vip!
Sorry about not including my version numbers too. On my production box I'm
using 1.8.3 (that's the debug from the original email). On my demo box I just
build I'm using 1.8 SVN-trunk-r309404 and that's what generated these logs. I'm
not sure if this is a chan_sip.c problem or
I'm starting a new project similar to a previous project where I used
OpenSER to front a bunch of Asterisk servers.
Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely
candidates.
I'm leaning towards OpenSIPS because it's in EPEL so I can install it with
yum. Also, because I
Hi,
I have been working on a project with asterisk and kamailio. I would
prefer using kamailio because i have personally met with the developers
and it has more active users and rapid developments. The developers are
also very friendly and helpful. And well open ser is not gone, the name
is
On Fri, Mar 4, 2011 at 11:28 AM, viswavardhanreddy karna
viswavardhanre...@gmail.com wrote:
Hi,
I mean when the cpu history is in idel and in busy state...
i have one more doubt that we are doing experiments on server
performance(only on software) it does not depends on hardware or
Does anybody have an answer to this?
- Original Message
From: Alfred Monticello ajmce...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Wed, March 2, 2011 9:59:20 PM
Subject: chan_skinny and Cisco 793X (7936) support in 1.8
Is there any way to make a Cisco 7936 conference
Skinny?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alfred
Monticello
Sent: Friday, March 04, 2011 2:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] chan_skinny and Cisco 793X (7936)
Hi,
We use Opensips and like the results. The forks are similar, docs from
one can help in the other. The opensips mailing list is monitored by
one of the main developers. He is even in the IRC chat in the mornings.
The docs are kept current on the opensips webpage. They like to change
I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco
spa8800, all them are internal lines.
1.- spa921, 401 ext
2.- spa921, 402 ext
3.- normal phone connected to spa8800 404 ext.
It had a very strange behavior when I was configuring call transfer and call
pickup.
These are
Hi,
We have worked out another approach for load testing:
- generate using sipp certain number of test calls and that go to PBX echo
server playing and receiving back pre-defined audio
- generate +1 test call, which also plays and receives back an audio file
Then we test the audio we
On Fri, Mar 4, 2011 at 12:50 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
Can you explain the main differences between Libpri 1.4.11 and 1.4.12 as
both seem to receive additions and patches ?
Do they target different asterisk versions ?
Can they both be considered as production-ready ?
1.4.12
Hi all,
I need to use asterisk to record all phonecall I have test using
mixmonitor to record a call.
Now I need to set the configure file to let asterisk auto record all
calls. I have searched many
document but still can not succeed. My version is 1.8beta and I prefer
using mixmonitor.
Regards!
For those interested, I have released a first version of jkSMS, which is
a simple package that lets cell phones text messages to asterisk.
Note it's not real SMS, it makes heavy use of email-to-sms gateways, but
it seems to work well. I have had the code running 12 hours, but
haven't found
On 3/4/2011 9:49 PM, John Wu wrote:
I need to use asterisk to record all phonecall I have test using
mixmonitor to record a call.
this is one way it can be done
make sure you have 'lame' installed.
- in your extensions.conf:
[global]
VSA=/var/spool/asterisk
[outbound-or-wherever-you-dial]
Once again, thanks for your reply. I had done some research already but
forget to include it in my previous email. I did find a bug that is
remarkably similar to the issues that I'm having. The bug number is 18674.
Thanks,
Mitch Johnson
Message: 8
Date: Fri, 04 Mar 2011 00:34:45 -0600
Hi
i want use the API on my asterisk 1.6, but i have a small problems :
In extension, i start it :
exten = _X.,3,AGI(My-Script.agi)
The perl agi file are started without problems
but i want get into this script a lot of variable:
Type (SIP or IAX)
src (from cdr)
but that's don't work:
Hi
I have two Asterisk Server:
The first server A, all phone are connected
The Second server B only route call to a lot of SIP supplier
the server A sent:
; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR
exten = _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW)
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