Displaying 20 results from an estimated 5000 matches similar to: "h323 oh323 g729 please help !"
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 Jul 06
3
H323 channel
Hello everybody,
my * box is connected to gnugk with H323 channel. If I call from an H323
EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio
start but noisy (scratch) , then became ok for callee (SIP EP) but still
scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323
EP and it's ok. And from now, it's also ok when H323 EP call SIP one's!
No
2004 Jul 22
1
Sip -> H323 using oh323 and G729
Hi All,
I have set up a box that will be used as follows:
SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server
192.168.1.5 192.168.1.50 192.168.1.80
Asterisk is running the latest CVS and oh323 driver.
The SIP phone is a Grandstream Budgetone 100.
I have everything setup and running with G.711 ALAW and ULAW and i'm able
to make calls through
2003 Aug 08
2
Re2: Problem -ATA-711-723-Oh323-Asterisk(BACKTRACK INFO]
Hello Michael,
Here is the BackTrace of the program which i forgot
to attach
BACKTRACE OF Asterisk -vvc
#0 0x42074d60 in _int_realloc () from
/lib/tls/libc.so.6
#1 0x420738c4 in realloc () from /lib/tls/libc.so.6
#2 0x47c7da89 in PAbstractArray::SetSize(int) () from
/home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5
#3 0x47c7cf4d in PContainer::SetMinSize(int) () from
2005 Mar 09
6
how to sip->h323 using asterisk-oh323-0.7.1
hello
i am using asterisk-oh323-0.7.1. i want to convert sip
call to h323 (h323 sjphone or h323 proxy). what could
be the best way for this. i am successfull in
converting h323->sip by using asterisk as gateway.
help required on sip->h323.
kamran
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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?
2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears;
I do not know if any had experience in using speex or
ilbc with IAX and got good results, because I am
facing a problem with GSM.
I am facing a noise problem when I am using GSM with
IAX trunk as following:
IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk
using GSM codec ---> Remote Asterisk Box ---> Digium
Card (FXO) to terminate the call to the destination
While no
2005 Jul 27
2
oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6
Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All
I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb
ram, with g729 for i686 , (fedora 1).
my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen
otherparty realtime voice , but other party geting sip party's voice 1 sec
later (not
2006 Nov 20
2
Recording g729
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=UTF-8" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Helvetica, Arial, sans-serif">Before ordering I want to be
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
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.
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
2003 Aug 07
2
Problem -ATA-711-723-Oh323-Asterisk
Hi List,
I am facing the reverse problem as stated here.I am
using ATA 186 to make
and recieve call to * through OH323 driver.
When I use G711 codec in the ATA to make call then
then as soon as i dial an
extension the * crashes with 'segmentation fault'.
But the same scenerio works fine when i use 723 codec
in the ATA .I can dial
the number and extension very well/(I have 723 support
in
2004 Jul 12
2
OH323 and G729
Dear All,
I have problem with new oh323 0.6.3a , I try use var OH323_OUTCODEC, but it don't work.
oh323 driver don't want connect to gateway with g729, it's work if I only use in oh323.conf one codec ( g729 ). If I enable 2 or more codecs - always in use other codec:
-- Executing SetVar("IAX2[4010@4010]/1", "OH323_OUTCODEC=g729a") in new stack
-- Executing
2008 Apr 01
1
g729 encoder/decoder
How does the g729 encoder/decoder count in regards to the total number
of licenses and how does it count an encoder/decoder? I looked on the
wiki and don't really see anything that explains it. In other words,
how do the calls below count (assume no reinvite)?
g729 phone calls into voicemail
g729 phone calls g711 phone
g729 phone calls other g729 phone
2005 Jan 05
0
One way audio [Asterisk + Innovaphone IP3000 + asterisk-oh323/h323]
Hello everybody,
I?ve been trying to solve a problem for several weeks now but it really
beats me.
There are several hard phones connected to an Innovaphone 3000 VoIP gateway.
On the other side I have a SIP softphone connected to Asterisk. The problem
I have is that on incoming calls (hardphones to softphone) I only have
outgoing audio (from soft to hardphone); everything is OK when I call the
2004 Aug 11
7
H323 call dropped when answered
Hi All.
I'm using RedHat 9
I configured the chan_h323 and asterisk from CVS.
This is the scenario SJ_lab_phone(sip) ---------------> Asterisk
-------------> H323 GK --------------> PSTN
I have tried all codec's and always the same result, the called phone
will ring without dropping for how ever I allow it to but as soon as it
is answered it immediately gets disconnected.
2006 Apr 19
1
Codec problem from SIP to H323
Hello.
I have a codec problem to send calls from a SIP device to a H323 gateway.
First I'll explain the scenario:
- Asterisk 1.2.1
- The SIP phone can use any codec I want.
- The H323 gateway can only use g729 (cause it's not under my
administration)
- SIP phone has g729 configured, so my asterisk doesn't need to "transcode"
(I don't have licences for g729)
- sip.conf
2007 Nov 28
4
G729/MOH Quality
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by calls coming from the PSTN via a Zaptel gateway. I am only using the
stock wav files and have not attempted to use much else so far.
I've ruled out timing issues on the system generating the MOH itself
(ztdummy on the PBX itself, our Zaptel gateway