Displaying 20 results from an estimated 2000 matches similar to: "Config file references (was g726 not working)"
2004 Mar 30
1
G726 not working ?
Hi,
I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of
this morning 3/30/04 of asterisk, zap and libpri.
The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced".
When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I
can see:
[format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
==
2005 Mar 21
2
G726-16 passthrough...
Hello,
I'm wondering if anyone has benn able to successfully get g726-16
passthrouhg to work? I am wanting to use this codec instead of g729 as
I'm running out of DSPs using a high complexity codec on the Ciscos. I
would think it would work just as g729 does, which has been working fine
for me, but it does not. G726-32 does work great however, but it's like
Asterisk doesn't
2005 Jun 02
2
asterisk sipura and g726 codec
With sipura (I tried this with both the 3000 and 841) set to prefer
the g726-32 codec, a call from the sipura to asterisk will use g726.
Asterisk sip.conf has:
disallow=all
allow=g726
allow=gsm
allow=alaw
When the call is from asterisk to the sipura, asterisk will not use
g726. It ends up using alaw. I usually use stable but I tried this
with head too, and same thing happens.
Anybody know how
2006 Apr 11
2
G726-40 required - Help!
Hi everybody,
A customer requires G726-40 with Asterisk... I know G726-32 is
pseudo-standard, but he definitely wants G726-40...
Is there any (easy) way, to integrate G726-40 into Asterisk? Has anyone
done this before? Any hints? Please help!
Due to a misunderstanding, my product manager already offered this to
the customer and now i do not know how to do it...
Thanks a lot in advance,
2010 Aug 02
6
Codec negotiation : expecting G726, getting G711a (alaw)
Hello list,
Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle.
Grandstream allows for 8 different codec specifications. I have defined
them as 4 x G726 & 4 x alaw.
Snom allow for 7 different codec specifications. I have defined them as
3 x G726 & 4 x G729.
The SIP peers are both defined as :
disallow=all
allow=g726
allow=alaw
allow=g729
allow=gsm
This is the
2007 Jul 20
1
ulaw to g726 conversion
I am able to use sox to convert audio files from ulaw to
wav (MS ADPCM), is there a way, using sox or another
command line tool, to convert them to g726 ?
( g726-32 since it is supported by * )
tia,
-baji.
--
2006 May 20
1
$1000USD for fix of Asterisk g726-32 codec
Hi All,
I am happy to offer $1000USD for the fix of the g726-32 in Asterisk.
What's wrong with it? It currently gives a very distorted sound as
though the gain is set to high. Lowering the gain on endpoints helps
but this is not a fix just a poor workaround. We require g726-32 to be
of the same quality as the Asterisk g711 implementation.
As the developer who fixes this issue you will
2014 Feb 11
0
g726 transcoding
Just checking the transcoding on our Asterisk boxes and I get the
following results.
I have the g726, ilbc and lpc10 formats and codecs enabled in 'make
menuselect' so I dont understand why its showing as no translation path.
Any ideas?
I am running certified-asterisk-11.2-cert2
Thanks
Gareth
> core show translation paths alaw
--- Translation paths SRC Codec "alaw"
2004 Dec 10
1
Doubts regarding g726 - 16 bits setup
Hi all,
I would like to make a call using the asterisk IAX
with g726 - 16 bits codec.
How could I configure it in the iax.conf file.
Do I need to modify the file like this?
.
.
disallow = all
allow = g72616k
.
.
I have tried it but it hasnĀ“t worked.
Thanks in advance and best regards
Guild
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2006 Jul 31
3
Soapbox
Hi all,
I thought y''all might be interested in seeing a newly released website
named Soapbox which was written in Rails.
Soapbox features reviews of products, businesses, services, and anything
else you can think of written by the people *you* care about.
http://soapboxit.com
Thanks!
Duff OMelia
--
Posted via http://www.ruby-forum.com/.
2004 Jul 09
1
RE: ATA 186, firmware SIP 3.1 and codec g.726 + now SIPURA SPA-2000
To me it's a error if I can't complete calls using the ATA configured to use
the g726 codec.
I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I
received NOTICES and WARNINGS, but I can't complete a call.
On a zap channel:
-- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack
-- Called 1/2217008
-- Zap/1-1 answered
2006 Mar 28
0
codec translation problem???
2006 Sep 05
3
terms.inner
Question:
I am trying to impliment a function in R that we use quite regularly in
Splus, and it fails due to a lack of the "terms.inner" function in R.
The substitute is?
Part question and part soapbox:
Why remove terms.inner from R? It's little used, but rather innocuous.
Mostly soapbox:
I figured it was no big deal, as I originally discovered the use of
terms.inner from
2006 Dec 06
1
FW: G.726 on Asterisk 1.4.0
Ok,
With everything restore on rtp.c, still I have no audio however the call is
not destroyed immediately as before.
I'm going to put a second Granstream box, and findout if between two boxes
this happen too.
I cannot believe that we cannot do 2 g726 on the same box at one time.
Carlos
-----Original Message-----
From: Carlos Alperin [mailto:calperin@senecacom.net]
Sent: Wednesday,
2005 Mar 20
1
I cannot use G711 (ulaw|alaw)
Dear all,
I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to,
I got the following error message:
Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect
attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with
our capability 0xfe02.
I do not understand why because my Asterisk box load these codecs properly!
Does somebody
2008 Jan 01
3
[1.4 + FreeBSD 6.2] Playing WAV PCM file?
Hello
Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...
www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
... I recorded a sample of my voice using XP's Sound Recorder, then
ran the following :
sox test_wav.wav -r
2004 Oct 07
1
Confused about NAT and Authentication with FWD
I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT.
The problem:
I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem.
Now if in the extension.conf file I have,
exten =>
2007 Jun 18
0
[Nut-upsuser] false alerts/shutdown
> driver.name: newhidups
[...]
> ups.status: OL CHRG
Euhm, there is no status 'CHRG' according to 'docs/new-drivers.txt'
folks. If you feel the need to create new status flags, at the very
least mention them in the section 'Status data' in this file. Unless you
document it somewhere, it's of no use to clients.
[soapbox]
Before doing so, ask yourself (and
2007 Jun 29
0
CHRG / DISCHRG status
(was: false alerts/shutdown)
2007/6/18, Arjen de Korte <nut+devel at de-korte.org>:
>
> > driver.name: newhidups
>
> [...]
>
> > ups.status: OL CHRG
>
> Euhm, there is no status 'CHRG' according to 'docs/new-drivers.txt'
> folks. If you feel the need to create new status flags, at the very
> least mention them in the section 'Status
2005 Jul 02
1
Sipura SPA2000 behind NAT
Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network
adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP:
___________ HOME _______________ ____OFFICE ____
SPA2000 <---> Linux Box <--> Asterisk Box
192.168.0.253 192.168.0.1 eth1 200.93.xxx.a
200.93.xxx.b eth0
My problem is when I try to call to any trunk or extention