Displaying 20 results from an estimated 20000 matches similar to: "VoIP carrier g729"
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
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
2006 Feb 17
4
one way / irratic voice over iax and g729
Hi All,
We are experiencing a a problem when running calls over IAX with g.729.
The call flow is as follows:
Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier
The first Asterisk system is running 1.2 and the second is running 1.0.
When using g726 from the handset all the way thru to Asterisk2(then 729
for the carrier leg) calls go thru fine, but when using g729, there
2020 Sep 24
2
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
2010 Sep 04
1
Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?
Hello,
We are in the process of debugging a voice quality issue for a client of
ours that is a VoIP services provider. The client uses a softphone that
runs on a pjsip stack.
When placing a call using the softphone, it negotiates the use of G729
codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP
packets with encoded G729 payload. VAD/DTX is enabled. We see that the
last
2006 Nov 20
2
Recording g729
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<title></title>
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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 18
1
h323 oh323 g729 please help !
Hello list,
I have many IP hardphone 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
I have this problem:
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 ( G729 ) only if I disable
2004 May 10
1
Testing IP phone (g729, g711) with Windows Messenger (g723, g711)
Hello, all.
I have some problem when testing my IP phone with Windows Messenger.
My IP phone supports such codecs as g729, g711.
And Windows Messenger supports red, g711, g723 as you know.
The problem comes up when testing with this sip.conf file. ([general] context displayed only)
===================================================================================
[general]
port=5060
2007 Sep 14
1
g729 on 1.4.10.1
I have a fresh 1.4.10.1 installation that appears to have a problem
with g729 pass-through. I can see the gateway in question sending an
INVITE using g729b. However, the Asterisk is only sending g711 in the
INVITE to my Polycom phone.
[gateway]
disallow=all
allow=g729
[phone]
disallow=all
allow=ulaw
allow=alaw
allow=g729
There's the codec configs for the gateway and the phone in question.
2007 May 08
1
G729 - Part cut
Hi all,
We are an ISP in Switzerland and we propose VoIP with Asterisk.
Everything works perfectly for all clients but one. In a conversation,
they have no sound during 2 to 8 seconds using the G729 codec (I didn't
make the test with G711).
The Client configuration is perfect (QoS and bandwidth management).
Do you know some issues with the G729 codec?
Thanks a lot for your comments,
Thomas
2003 Oct 03
1
Budgettone + G729
hi there ..
I asked sometime ago regarding getting a Budgettone
working with Asterisk over G729.
My system is quite simple, Asterisk server with 1 G 729 license
installed, and 10 Grandstream phones. Only one of them needs
G729, because it's on a remote link via an ADSL bridge. The
rest run happily on G711 on a local network.
I added the lines
disallow=all
allow=g729
to the sip.conf entry
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
2007 Sep 26
2
My G729 problem re-visited
Ok, I built a test system to duplicate my problem and provide myself
a platform that I can mess around with to try and break any features.
My problem is G729 pass-through from a gateway to a phone. I think
I even have transcoding working, which makes me more confused on
what's wrong with my pass-through. It must be a configuration issue.
The basics...
*CLI> core show version
Asterisk
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
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
2004 Jun 18
2
C7960 g729 question
I have multiple voiceage g729 licenses installed on a RH9 box, and have
a remote C7960 configured to use it (low bandwidth). In calls like:
Remote C7960 -> g729 -> asterisk -> g711 -> C7960
the audio is oftentimes rather choppy. Changing the remote 7960 to use
g711 seems to eliminate/reduce the choppyness. Any ideas on what might
be behind this?
2006 Dec 03
1
G729 Passthru?
I have a SIP carrier which accepts only G729 connections from my
Asterisk server. If all the server does is Dial() (out) two legs of a
call which are natively bridged, with no processing the media (and no
DTMF detection, etc), do I need to install a G729 codec of my own? All
the media from each leg connected to the other is already encoded into
G729 by the SIP carrier from which it's coming
2011 Jun 13
5
No audio after a reinvite changing codec
Hi all,
we have a problem with a reinvite sent by our SIP provider to change audio
codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE and 200 OK) we
have the following codec situation:
UAC ASTERISK UAS | ASTERISK UAC
PROVIDER
g711 <----------------------> g711