Sorry for the top-post...
If you do a core show application AddQueueMember from the cli, you'll see the
option I was referring to.
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
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
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 Darnellmattdarn...@gmail.com wrote:
Warren,
I tried using
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
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=yes
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
_
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
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 8Khz
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
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 in
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
--
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
*
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
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
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
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 see
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 Fri, Oct 15, 2010 at 11:02 AM, Danny Nicholas da...@debsinc.com 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
On Fri, Oct 15, 2010 at 10:29 AM, Steve Edwards
asterisk@sedwards.com 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
-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
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 8Khz
On Fri, Oct 15, 2010 at 11:20 AM, Danny Nicholas da...@debsinc.com 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
[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,
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
On Fri, Oct 15, 2010 at 11:50 AM, Jeff LaCoursiere j...@sunfone.com wrote:
snipped
(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.
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
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
--
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 how can
On Fri, Oct 15, 2010 at 8:53 AM, Michelle Dupuis mdup...@ocg.ca 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
-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
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 criminal
On Fri, Oct 15, 2010 at 9:35 AM, Danny Nicholas da...@debsinc.com 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
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen
leif.mad...@asteriskdocs.org 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
On Fri, Oct 15, 2010 at 11:07 AM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de 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
2010/10/15 Matt Darnell mattdarn...@gmail.com:
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen
leif.mad...@asteriskdocs.org 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
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
On Fri, Oct 15, 2010 at 1:44 PM, Michelle Dupuis mdup...@ocg.ca 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
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 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
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 another file
On Fri, Oct 15, 2010 at 9:55 AM, Jared Geiger compuw...@gmail.com 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
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
to
[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
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
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