I haven't heard if this fixed it yet. However I was seeing the echo
cancelers loaded before so I never realized I'd have to do this. Its a
FreePBX install also so I checked all the include files and didn't see a
reference to these values anywhere.
Thanks everyone for the input, I should know soon
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
> Fleming
>
> There were some comments in other replies about your files being 'quiet'
> (low average volume level)... this won't help your situation at all,
> because it means that any artifacts caused by resampling and
>
On Fri, Oct 15, 2010 at 06:22:07PM +0200, Zarko Zivanovic wrote:
> We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this
> natted network.
>
> We have the issue with calls to these SIP phones - no audio.
Tell us more about your settings. I have a GXP2000 behing NAT connected
On Fri, Oct 15, 2010 at 9:55 AM, Jared Geiger wrote:
> I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a
> full reformat and recompile) and I started getting echo over the PRI.
>
I did an update on a server last year, had the same problem. I needed
to explicitly set echocancel=ye
On 10/15/2010 10:33 AM, Jared Geiger wrote:
> This might be my problem?***
> [r...@voice ~]# grep -E "^echo" /etc/asterisk/chan_dahdi.conf
> *
> *So I added this under [channels]:
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=800*
>
>
Most likely (unless you were including anoth
On 10/15/2010 04:00 AM, Karsten Wemheuer wrote:
> I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with
> dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an
> older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN
> card. The system starts without any
Hi!
> Trunking only reduces overhead after 4+ calls, so that shouldn't help
> either. (Since this occurs at 2 calls)
Trunking requires a timing source, and you might have trouble with your
timing, that is why I suggested this (and because you did not tell us
wether you have trunking enabled or
On Fri, Oct 15, 2010 at 1:44 PM, Michelle Dupuis wrote:
> Jitterbuffer affects inbound audio only, not outbound (the other side hears
> the choppiness) so I don't think that will help/
>
If your problems with audio are at the far end, I don't expect there
is much you can do. Try a different code
Jitterbuffer affects inbound audio only, not outbound (the other side hears the
choppiness) so I don't think that will help/
Trunking only reduces overhead after 4+ calls, so that shouldn't help either.
(Since this occurs at 2 calls)
I can't wireshark the other end since the other end is my IT
2010/10/15 Matt Darnell :
> On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen
> 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 alternative for obsolete call-limi
On Fri, Oct 15, 2010 at 11:07 AM, Philipp von Klitzing
wrote:
> - turn off IAX trunking mode
>
I would disagree, you want to enable trunking with multiple call. It
will reduce patch overhead, leading to less bandwidth.
OP could enable jitterbuffer, if not already enabled.
--
Paul Belanger | dC
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen
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 alternative for obsolete call-limit option in 1.6/1.8?
>
> The correc
On Fri, Oct 15, 2010 at 9:35 AM, Danny Nicholas wrote:
> Don't know if this will make "acceptable" GSM files, but should help with
> the WAV ones.
>
Are you using GSM to talk to an ITSP (the idea of banking voice calls going
across the internet makes me cringe)? If not, what are you using GSM
On Fri, 2010-10-15 at 07:29 -0700, Steve Edwards wrote:
> On Thu, 14 Oct 2010, bruce bruce wrote:
>
> > But it also sickens me at how badly Asterisk is made to not cope with
> > situations like this and worse than that is FreePBX.
>
> Kind of like blaming the gun manufacturer instead of the crim
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Friday, October 15, 2010 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP - no audio behind n
On Fri, Oct 15, 2010 at 8:53 AM, Michelle Dupuis wrote:
When a single call is up, call quality is fine. When a second call is up,
> outbound audio is immediately choppy. We're using ulaw, and confirmed that
> traffic with 2 calls is <175kbps in/out. (IAX connection out)
>
> Asterisk doesn't re
On Friday 15 Oct 2010, Zarko Zivanovic wrote:
> Hello,
>
> We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this
> natted network.
>
> We have the issue with calls to these SIP phones - no audio.
>
> It is probably the problem with port forwarding on router - but I am not
> sure
On Fri, Oct 15, 2010 at 06:22:07PM +0200, Zarko Zivanovic wrote:
>We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this
>natted network.
The simplest solution will be to stick another Asterisk box inside the
NAT and tunnel IAX or SIP over a VPN.
R
--
___
Hello,
We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this
natted network.
We have the issue with calls to these SIP phones - no audio.
It is probably the problem with port forwarding on router - but I am not
sure how can I forward same sip ports (5004 to 5100) to two
On Fri, Oct 15, 2010 at 11:50 AM, Jeff LaCoursiere wrote:
>
>
>> (BTW Sierra Leone is in West Africa, not the Middle East.)
>>
>
> True ;) Most of the calls were Iraq, UAE, Lebanon... Found another one
> today that was 2.5 DAYS long to Chile. Bizarre.
>
> j
>
Not bizarre at all. You being in
We took a pretty nasty hit one time, a system administrator didnt listen to
us about changing the passwords. Luckily they took part of the blame in
that, and we split the 1800$ it cost us in half. We could have changed
them, and she didnt change them, so we were both at fault.
Like said previous
I would like to know about that Chile destination.
always start here: http://www.spamhaus.org/drop/
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about
On Fri, 2010-10-15 at 11:20 -0400, Steve Totaro wrote:
> This is nothing new. Trunk to trunk transfers and other exploits
> could be used on old school phone systems to do the same thing.
>
> I would start with getting the current balance, if over $10k call the
> FBI, call them anyways, it could
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Friday, October 15, 2010 10:25 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] drop dead fix
On 10/15/2010 08:59 AM, Da
[r...@voice ~]# cat /etc/dahdi/system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Fri Sep 24 21:44:03 2010
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahd
On Fri, Oct 15, 2010 at 11:20 AM, Danny Nicholas 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 pay gr
On 10/15/2010 08:59 AM, Danny Nicholas wrote:
> Hello list,
>
> I am about to have to dump Asterisk in favor of some other
> VOIP/PBX solution; the reason? I have 304 voice prompts recorded as
> 22Khz wav format files that sound like crumpling paper whenever I
> convert them to the
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Deneen
Sent: Friday, October 15, 2010 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] drop dead fix
On Fri,
So I'm in a situation where I want to consolidate cdr logs. My general
idea is to use cdr_mysql for this.
I know I can do things like
Set(CDR(userfield)=hostname)
And I can hardcode the hostname for the dialplan on each system.
But what I'd really like to do is have this dynamic, so I can use th
On Fri, Oct 15, 2010 at 10:29 AM, Steve Edwards
wrote:
> On Thu, 14 Oct 2010, bruce bruce wrote:
>
>> But it also sickens me at how badly Asterisk is made to not cope with
>> situations like this and worse than that is FreePBX.
>
> Kind of like blaming the gun manufacturer instead of the criminal
On Fri, Oct 15, 2010 at 11:02 AM, Danny Nicholas 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 phones?
--
Hi!
> Can someone suggest where to look? Could this be the ITSP?
- turn off IAX trunking mode
- test with SIP to find if it IAX causing the trouble
- capture the RTP traffice on the other side and let wireshark have a
look at that stats (loss, jitter)
Philipp
--
__
On 10/15/2010 08:55 AM, Jared Geiger wrote:
> I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a
> full reformat and recompile) and I started getting echo over the PRI.
>
> I've tried the default settings for echo in the system.conf file as
> well as I've compiled OSLEC to try and
On Fri, Oct 15, 2010 at 10:41 AM, Danny Nicholas wrote:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
> Sent: Friday, October 15, 2010 9:21 AM
> To: Asterisk Users Mailing List - Non-Commer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent: Friday, October 15, 2010 10:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] drop dead fix
On
On Fri, 15 Oct 2010, Danny Nicholas wrote:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
> Henderson
> Sent: Friday, October 15, 2010 9:18 AM
> To: Asterisk Users Mailing List - Non-Commercial Dis
We have a small office installation running over a cable modem. (8M down, 500k
up confirmed with numerous speed test sites)
When a single call is up, call quality is fine. When a second call is up,
outbound audio is immediately choppy. We're using ulaw, and confirmed that
traffic with 2 call
For future I would highly recommend to have at least fail2ban installed.
This way sipvicous IPs will be blocked instantly before they could create
any damage. Also I prefer to limit International calling to only certain
limit, e.g. only for $10 per account, but this depends upon how your
business d
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, October 15, 2010 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] drop dead fix
On Fri
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon
Henderson
Sent: Friday, October 15, 2010 9:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] drop dead fix
On
You want to pay attention the low-pass and high-pass filter A
step conversion will help you see the issues. Go halfway first and
look for the change and adjust your filter.
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
I never had this problem, and this is certainly not asterisk's fault.
Probably your conversion is not good. Can you email me a file and I'll do
conversion on my end, and if sounds good, let you know how I did it. Then a
script can be written to convert them all.
Zeeshan A Zakaria
--
www.ilovetovo
On Thu, 14 Oct 2010, bruce bruce wrote:
> But it also sickens me at how badly Asterisk is made to not cope with
> situations like this and worse than that is FreePBX.
Kind of like blaming the gun manufacturer instead of the criminal with
their finger on the trigger?
Is there some gaping hole i
On Fri, 15 Oct 2010, Danny Nicholas wrote:
I am about to have to dump Asterisk in favor of some other
VOIP/PBX solution; the reason? I have 304 voice prompts recorded as
22Khz wav format files that sound like crumpling paper whenever I
convert them to the 8Khz wav/gsm format re
On Fri, 15 Oct 2010, Danny Nicholas wrote:
> Hello list,
>
> I am about to have to dump Asterisk in favor of some other
> VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz
> wav format files that sound like crumpling paper whenever I convert them to
> the 8Kh
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder
Sent: Friday, October 15, 2010 9:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] drop dead fix
On 10/15/2010 09:59 AM, Danny Nicholas wrote:
pbx$ man sox
allpass frequency[k] width[h|k|o|q]
Apply a two-pole all-pass filter with central frequency
(in Hz) frequency, and filter-width width. An all-
pass filter changes the audio's frequency to phase
relationship without changing its frequency to amplitude
On 10/15/2010 09:59 AM, Danny Nicholas wrote:
Hello list,
I am about to have to dump Asterisk in favor of some
other VOIP/PBX solution; the reason? I have 304 voice prompts
recorded as 22Khz wav format files that sound like crumpling paper
whenever I convert them to the 8Khz
Hi,
We have realtime queues, and I can't figure how the device state matters,
because when a user takes a call, is stat is "Not in use".
We now use GROUP_COUNT() to check if the peer has a call or not...
On 15 October 2010 12:21, Leif Madsen wrote:
> On 10-10-15 04:10 AM, Сикорский Сергей wrot
Hello list,
I am about to have to dump Asterisk in favor of some other
VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz
wav format files that sound like crumpling paper whenever I convert them to
the 8Khz wav/gsm format required by Asterisk. I was consider
I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a
full reformat and recompile) and I started getting echo over the PRI.
I've tried the default settings for echo in the system.conf file as
well as I've compiled OSLEC to try and see if thats any better.
I'm not sure what to try nex
On Fri, Oct 15, 2010 at 9:38 AM, wrote:
> I was redirected here from the -dev group since meetme depends on dahdi.
> I'm still a newbie in asterisk programming.
>
> I'm attempting to do some custom modifications to the meetme application. I
> got a WARNING[10419]: pbx.c:3680 pbx_extension_helper:
I was redirected here from the -dev group since meetme depends on dahdi.
I'm still a newbie in asterisk programming.
I'm attempting to do some custom modifications to the meetme application. I got a WARNING[10419]: pbx.c:3680 pbx_extension_helper: No application 'Meetme' for extension (inmate_p
On 10-10-15 07:20 AM, Leif Madsen wrote:
> On 10-10-14 10:49 PM, bruce bruce wrote:
>> Unfortunately, probably there is no one you can complain to. But it also
>> sickens me at how badly Asterisk is made to not cope with situations
>> like this and worse than that is FreePBX.
>
> How is password po
Auditing is an important process of any system. Automatic auditing
against CDRs is not that hard and phone calls that happen at 1am are
easy to see. I would suggest a CRON job to email all calls that
happen outside normal business hours to the owner of the phone
system.
~
Andrew "lathama" Lath
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 alternative for obsolete call-limit option in 1.6/1.8?
The correct answer is to use ringinuse=no in queues.conf and callcounter
On 10-10-14 10:49 PM, bruce bruce wrote:
> Unfortunately, probably there is no one you can complain to. But it also
> sickens me at how badly Asterisk is made to not cope with situations
> like this and worse than that is FreePBX.
How is password policy an Asterisk issue? The solution to the probl
Hello,
I'm having a very similar issue with dahdi 2.3.0.1 / 2.6.32 (and others
confirmed the occurence with same software revisions, same kind of old
hardware - P3, P4, different HFC hardware). You can look at my last
report on loosely related debian bug #598886.
http://bugs.debian.org/cgi-bin/bu
My problem with call drops on PRI, followed by 'FRAME_CONTROL (8)' was
successfully solved. The issue was in callprogress=yes option in
chan_dahdi.conf. It seems that I wrong when I wrote configuration file
and confused it with usecallingpres option.
On 30.09.2010 16:57, Захаров Антон wrote:
On Thu, Oct 14, 2010 at 11:57:24AM -0500, Carlos Chavez wrote:
> > But what ports did you open? Only sip or also the RTP ports?
> >
> I opened SIP and RTP, after that I put the server on the DMZ but I
> still get no audio on the external phone.
>
> My problem is that we do not adminis
Hi,
I setup an asterisk system (asterisk 1.8-rc3, dahdi-linux-2.4.0 with
dahdi-extra from Tzafrirs git, kernel 2.6.35.4). The hardware is an
older pc system with Celeron CPU (2.5 GHz) with a Beronet BN4S0 ISDN
card. The system starts without any errors.
I discovered a severe issue. The kernel pan
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 alternative for obsolete call-limit option in 1.6/1.8?
>
> Thanks,
> --Warren Selby
>
> On Oct 14, 2010, at 11:36 PM, Matt Darnell wrote:
>
>> Warren,
>>
>> I tried using AddQueue
> You'll also need to make sure you're properly reporting device state to
> asterisk. I think this means you need to set a call-limit for each sip peer
> that you want to monitor in sip.conf (we use 25 so there are no accidental
> limits actually applied), and setup hints in your extensions.conf
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