Displaying 20 results from an estimated 1100 matches similar to: "G.729 disappears from h.323 codecs. Help, please!"
2003 May 28
0
calls between SIP and H.323 clients
Hello all,
It's me again.
I would like play with calls between a H.323 client and a SIP client
through * inside my LAN.
For that,
on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk;
on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I
register into * with a username, no password. The 3 files oh323.conf,
sip.conf, extensions.conf are in attachment.
In the same
2006 Jun 22
1
PCI or MiniPCI Hardware DSP for G.729, G.723.1 and/or GSM
Hi to all,
we are searching for a hardware based DSP solution for use
with Asterisk based on PCI or MiniPCI to reduce main processor
load and to use embedded boards with Digium E1/T1 cards like
TE410P.
does anyone know about any manufactorer of those cards or someone
who is able to develop/build such cards?
Specifications:
PCI or MiniPCI
up to 120 concurrent transcodings
Codecs: G.729/G.729A or
2004 Aug 13
1
OH.323 Dialout Problem
Hi,
I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular
phone. Asterisk configuration is listed below. When I attempt to place a
H.323 call, I receive the following errors:
- Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20")
in new stack
Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path
exists
2004 Jul 29
0
G.729 between Zap and SIP
Hi,
I have licensed the digium G.729A codec. But for some reason incoming and
outgoing calls will ALWAYS use G.711a. When I force my phone to only accept
G.729 then an incoming call from ZAP goes straight to my voicemailbox as the
phone doesn't accept the codec Asterisk wants, even if I force it in
sip.conf.
Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ?
The
2004 Jul 30
0
G.729 <-> ZAP ?
Hi,
I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card.
Incoming calls and outgoing calls between my cisco and my SIP phone works
fine on G.729. Recording messages in the asterisk voice-mailbox also works
fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have
licensed the digium G.729A codec.
When I connect my ISDN PRI to my Zap card and I call
2004 Oct 05
0
How to force G.729 in H.323 calls
Sirs,
I have the following problem:
I'm working with H.323 and in my h323.conf file (within the general
context) declare
disallow=all
allow=G729
but CLI shows:
*CLI> h.323 show codecs
Allowed Codecs:
Table:
G.729A{sw} <1>
G.729{sw} <2>
Set:
0:
0:
G.729A{sw} <1>
G.729{sw} <2>
And my asterisk first tries to negotiate G.729A
2005 Sep 04
0
Open G.729 / G.723.1 update, fixed memory leak
A new release of the open source G.729 patch has been issued.
The new URL is:
http://www.readytechnology.co.uk/open/ipp-codecs
The memory leak in codec_g729 is now fixed. This was due to a
problem in a section of code copied from the Intel example. Thanks
to those who assisted in locating this bug. If you are still running
the old version of the codec, your Asterisk process will run out
2003 Jul 10
2
OH323 + G729 + Go2Call
hi ..
i've just installed and licensed an instance of the G729 codec.
I am trying to connect through asterisk to Go2Call server ..
According to their info it involves dialling extension 729 on
voip01.go2call.com, to get the IVR.
my extensions.conf shows :
exten => s,2,Dial(OH323/h323:729@216.52.153.206)
which I think is correct, I have G729 enabled in the OH323.conf
file and it seems to
2003 Oct 23
1
Problems with OH323/codecs
On oh323.conf I have:
codec=G711U
frames=20
But while connecting it gives me in log:
? 1:18.636 ? ? ? ? ?H225 Caller:8111de8 H245 ? ?Capability merge result:
? Table:
? ? G.723.1(5.3k){hw} <1>
? Set:
? ? 0:
? ? ? 0:
? ? ? ? G.723.1(5.3k){hw} <1>
Which I don't have, so the connection is dropped. Any known solutions? (remote
side has g711 u-Law)
--
Witold Kr?cicki (adasi) adasi
2004 Dec 01
0
Diagnosing codecs
Hello,
I am trying a setup that is the following:
SIP Phone (Zultys) --> Asterisk ---> H.323 GK (Cisco) ----> PSTN
Any calls from H.323 GW through GK goes to PSTN, no problem.
SIP Phone registers to Asterisk, and calling to Voice Mail, No Problem.
SIP Phone to PSTN, rings normally, on the PSTN, then connects when the PSTN
phone picks up, no audio on both directions.
PSTN GW support
2005 May 11
0
Vegastream assistance?
I wonder if anyone can help me?
Am trying to terminate to H323 Vegastream. I'm using OH323 with little
success.
I can dial out and answer but voip end just keepings ringing and ringing.
Thanks for any help.
Neil
Config file:
[general]
listenAddress=ALL
listenPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=20000
fastStart=no
h245Tunnelling=no
h245inSetup=no
jitterMax=100
2005 Feb 01
1
Re: Asterisk-Users Digest, Vol 6, Issue 325
> Message: 1
> Date: Fri, 21 Jan 2005 17:38:27 -0600
> From: "Henry Devito" <hdevito@qwest.net>
> Subject: [Asterisk-Users] SPA-2000
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
> Message-ID: <000d01c50012$4ea49f30$4300000a@homeacxa7jw2xn>
> Content-Type: text/plain; format=flowed;
2009 Feb 23
3
GSM codec is a good choice ???
Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
with GSM sound files.
The problem is I have IP phones Utopix HyperPhone 202 which support
only G.729a/u and G.723.1 high/low, but not GSM.
If I choose G.729A the "pass-throu" calls among users are OK, but
Asterisk can't transcode GSM to G.729A in voicemail calls.
My company doesn'y want to pay for a G.729
1999 Aug 02
2
HTML Output from R
Task: To generate HTML output from R
Details: I am trying to serve up HTML output from R. That is analyses or
table of data from R saved as HTML output with formatted tables etc. This
file is then called in a CGI script to output to user browser. The CGI
script inspired by Mark J. Ray reads the HTML output, formats the header
and footer and any graphic output if necessary.
My question is: Has
2005 May 25
0
oh323 problems - Solved
For the benefit of everyone, having H323 Configuration problem due to
H245 Tunnel, check the h323 Config embeded at the end. Comment the
offending line as under:
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
-----Original Message-----
From: Tola Ogunsan [mailto:tolaniye@hotmail.com]
Sent: Wednesday, May 25, 2005 1:03 PM
To: Kanuri, Seshu (Company IT)
Subject: RE: oh323 problems
2003 Nov 06
3
which channel format number is right?
Hi all,
if i enter a "show codecs" at cli * response with:
1 (1 << 0) G.723.1
2 (1 << 1) GSM
4 (1 << 2) G.711 u-law
8 (1 << 3) G.711 A-law
16 (1 << 4) MPEG-2 layer 3
32 (1 << 5) ADPCM
64 (1 << 6) 16 bit Signed Linear PCM
128 (1 << 7) LPC10
2005 Mar 24
1
Optimized Codecs for Blackfin DSP
Dear Jean,
The source code for G.729 can be download from ITU for free. Also,
some developer can do yourself as open source G.729 codec without any
help. In this case each who use this codec which source code is free
and open source must pay, but not to the developer.
Best Regards,
Miroslav Nachev
JMV> Le jeudi 24 mars 2005 ? 10:08 +0000, John Villar a ?crit :
>>
2007 Oct 10
3
G729a codecs + Asterisk 1.4.11
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Good Morning,
Any help would be grateful to help me understanding what's wrong...
I have bought 2 g729a licenses to digium and I would like to have them works...
My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors)
so I have downloaded the
2004 Dec 07
3
:: Migrating to 1.0.3 => Attention. ::
Hello list ,
I?d like to announce possible problems with migrating any version prior to
1.0.2 to 1.0.3.
Pay attention :
1. Codecs
Codecs names/description have been changed .
For example :
versions <= 1.0.2
voip*CLI> show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
1 (1 << 0)
2005 Mar 24
0
Optimized Codecs for Blackfin DSP
Le jeudi 24 mars 2005 ? 10:08 +0000, John Villar a ?crit :
> Jean-Marc, that's not the definition of OpenSource, that's the
> definition of "Free Software" (Libre!=Gratis)
Open-Source and Free Software are basically doing the same thing for
different reasons. If something isn't free software, it's not
open-source either. There may be exceptions, but I have yet to