Displaying 20 results from an estimated 2000 matches similar to: "Testing IP phone (g729, g711) with Windows Messenger (g723, g711)"
2009 May 19
1
Alternative to Adobe Audition 3 for G723 > G711 uLaw ? (old Cool Edit Pro)
Can anyone recommend a codec pack with G723 for use under Vista? I have
G723 prompts (about 70 prompts totaling 1MB) needing to be converted to
G711 uLaw.
I tried Audacity but it doesn't have G723 codecs. I tired some google
found adware free tools and websites with no success in converting G723.
It does appear the old Cool Edit (now Adobe Audition 3.0 for $349USD)
can do it -jason
2006 Mar 31
1
transcoding g723 or g729 on asterisk
Kai,
Thank you for the reply.
I didn't want to bother the list too much. However, after reading I discover
I don?t have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this transcoding
done.
My set up
>From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same
asterisk) g711 to chan_ss7] -----> [pstn]
And vice versa.
I
2004 Jul 27
2
g729 + GSM + g723
Folks!
We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found.
Here is the config I have used:
-------------------------------
Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2
User1 is in USA on Broadband Cable
User2 is in India on 64Kbps ISDN Line
User1 using SIPURA SPA 2000
user2 using Xten professsional(X-pro)
2004 Apr 20
1
h323 and oh323 g711 to g729 please help
Hello list,
I have many IP hardphones like Siemens 300 basic ( old ) , cisco
ata.. etc
I need: G711 from old phones must be convert to G729 via asterisk and
send to provider ( G729 from digium )
I have this problems:
oh323 (last version):
-------------
asterisk work with this driver ok for old phones, if I only
faststart=no . But problem with codec , asterisk can speak with
provider (
2005 Mar 28
1
H323: g711-g729 transcoding
I have a connect to * via H.323/g711 from device A and want to connect
to B which want for H.323/g729
h323.conf contains
disallow=all
allow=alaw
allow=g729
but outgoing faststart/TCS contains only g711 (from h323_request(format)
i think) and so no codec negotiation and no voice.
Howto run up g711/H323 -> * -> g729/H323
PS intel's g729 was used. ast 1.0.3-6
PPS
stupid
-
2004 Apr 25
3
Grandstream Budgetone G723, G729 or any compression
Hi, does anybody made G723 or G729 to work with a GrandStream Phone ? I've
a Cisco here and it works fine with G723, but not with my asterisk. The
bandwitdh is very important, since we will have our extensions at home. I
know that I have what I pay, but the phone works with cisco.
Trying to use G723 or G729 Asterisk says no codec available.
Does anybody have it working with any compression
2020 Sep 23
0
Negotiates g729 but RTP contains g711
Hi,
We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip
2011 May 05
1
asterisk for g729 to g711
Hi,
Does anyone know if Asterisk is a good tool to be used for a large quantity
of g711 and g729 transcoding?
What is the best alternative for that?
--
Woody Dickson
woodydickson at gmail.com <woody.dickson at gmail.com>
US and Worldwide Termination
============ Contact me for the following offering ============
USA Onnet - 0.0049/min
USA Offnet - 0.011/min
USA Mobile starting
2005 Aug 06
0
g729 pass-thru for sip provider and g711 ulaw for conference and voicemail
Hello,
I'd like to use g729 pass-thru when I dial out to a sip provider from my
IP phone but because I have no license for g729 I'd like to use g711 ulaw
for asterisk voicemail, conference bridge and other services.
When I set in [general] section of sip.conf the following:
disalow=all
allow=g729
allow=ulaw
the g279 pass-thru works fine with my SIP provider but
when I call the
2004 Oct 01
1
Please, send me g723 & g729, pls
Somebody must have! Please, send me a g723 and/or g729 (for Asterisk) to
pisac@hotmail.com (antispam subject: codec)
Thanks, thanks, thanks...
:-)
2020 Sep 25
0
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2003 Nov 19
2
g723 to g723 SIP call - warning message
Hi,
I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.
Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.
[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new stack
--
2020 Sep 25
0
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2005 Jan 30
4
detailed asterisk howto
Hi, all:
I am a newbie to the asterisk and its architecture. :(
After reading some help in the tarball of Asterisk, I am
still in the mess. So I want to know where I can find a
detailed explanation of the Asterisk which including the
Architecture, Install, Configure, usage example document.
Maybe what I want is too much, after all it is a open
project, not commercial product. If I want to get
2004 Jul 19
0
Setup for Go2call ? Or any SIP provider using phonejack or linejack g729 g723
Hi, does anyone have the setup for go2call ?
I have digium boards and quicknet linejacks and phonejacks.
The cards work fine in asterisk without the g729 or g723.1 for the
phonejack.
I will like to do SIP origination using the codec in the phonejack and
linejack g729 or g723 and send the calls to go2call.
Anyone has the setup for this ? Or similar setup to a SIP provider using
g729 or g723
2004 Apr 13
3
CallerID in Australia
Well, Once upon a time, I had problems receiving callerid, and then one
day, Mark was logged into my asterisk box helping with something else,
and I asked him about this, and he showed me a nice tweak to some source
file that made it work.
Some time later, I must have done hundreds of CVS updates, and along the
way, lost the above patch (one liner) and so callerID hasn't worked for
a long
2020 Sep 22
2
Negotiates g729 but RTP contains g711
Hi,
We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip
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
2005 Feb 20
8
Simulated dialtone like in other PBX
Guys..
Im new to asterisk but is it possible to simulate a dialtone for example, in
other PBX when you pick up the phone you can hear a certain dialup, which is
the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is
this possible?
__________________________________________________________________
Anton Krall