Displaying 20 results from an estimated 200 matches similar to: "Re: Problem with Openh323 channel driver"
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
2004 May 18
0
problems with asterisk-oh323
Hello,
I've been trying to send traffic to a Cisco Call Manager 3.2, but with
no luck.
Here's whats happening:
* Call gets to CCM
* Call gets to the gateway
* Rings a couple times on destiny
* Call gets hungup.
On the CCM I get the following error: MediaManager - ERROR
wait_AuConnectErrorInd
On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not
available)
On asterisk:
2003 Jul 11
8
G729 codec problems
Hi I seem to have installed the G729 codec properly
but can't seem to get it to work ..
in the Asterisk startup I see ..
[codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator)
== Detected 1 licensed G.729 transcoders
WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator
'g729tolinb' does not produce sample frames.
== Registered translator
2005 Mar 04
1
Openphone implementation of Speex Codec's descriptions help
Would someone kindly share some definition into the following?
Openphone version 1.91 includes dual sets of Speex codec's starting with:
SpeexNarrow-5.95k{sw}
SpeexNarrow-5.95k{Xiph}
Through
SpeexNarrow-18.2k{sw}
SpeexNarrow-18.2k{Xiph}
I do not understand what the differences are between {sw} & {Xiph} given the
same bit rate for both?
Are all of these Narrow or Wide or Ultrawide
2003 Sep 12
2
problem with * and Howlink CL-100 ip phone
I'm trying to use a Howlink CL-100 ip phone with *
It's h323 phone with very limited protocol support. But it's enough that I
can use it to dial netmeeting client and artisoft pbx just fine.
When I try to dial my * with it using either chan_h323 or oh323, it seems
to fail on negotiating H245. Maybe this phone doesn't support it?
I've used all different versions of
2003 Nov 27
6
Help for oh323
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the extention.conf like this:
exten => _87.,1,Dial(OH323/16.52.153.206)
oh323 dials out (I can ring a
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
2004 Oct 08
0
problems with asterisk-oh323-0.6.3b
Hi guys,
I've been trying to update my chan_oh323 from 6.1 to 6.3b.
I built asterisk from cvs-head on the date Micheal said he made it
compatible, pwlib-1.6.6 and openh323-1.13.5 (both with nothing more than
the ./configure, make, well aplied patch on openh323)
When I start * with my normal config I get this:
[chan_oh323.so] => (OpenH323 Channel Driver)
== Parsing
2006 Nov 23
1
asterisk 1.4 chan_h323, help please...
Hi,
My configuration is SipPhone<-->*1<--->*2.
My asterisk version is 1.4beta3.
I installed pwlib,openh323,chan_h323.
When i call from
SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2,
there is no audio.
Using 'rtp debug', I can see that rtp packets are
being received.
Rtp packets are being exchanged.
I also tested chan_ooh323, but to fail.
Can anyone recommand best
2004 Aug 06
0
Re: Please confirm your message
>From: speex-dev@xiph.org
>To: wolfkharl@hotmail.com
>Subject: Please confirm your message
>Date: Fri, 06 Jun 2003 07:08:06 -0400
>
>Hello, this is the mailing list anti-spam filter at Xiph.Org.
>We need you to confirm your e-mail message with the subject of
>"Adaptivity".
>
>Please send a message to the following address, or simply use your
>mailer's
2005 May 30
0
IAX2 to H323
Hi all,
I'm using following software and equipment and I have very strange behavior:
Asterisk CVS-NHEAD-05/30/05-16:42:41
H323 gatekeeper - GnuGK 2.2.2
IAX2 client - Firefly 1.9.8 build 3945
H323 client - SJPhone Build 1.50.271d
H323 gateway - Welltech Wellgate 3504A
When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected.
When I dial from SJPhone (H323) ->
2009 Jul 06
1
TOSHARG-DomainMember.xml translate finish and some bug found
Now, TOSHARG-DomainMember.xml translate to Japanese finished.
and Some bug found.
<procedure>
<title>Server Manager Account Machine Account Management</title>
-------Domain?
<step><para>
From the menu select <guimenu>Computer</guimenu>.
</para></step>
When the user elects to make the
2005 Feb 14
0
H323 no sound
Could you help me with this problem? When I call H323 gateway there is no
sound in both ways.
Here is h323 debug:
----- begin ------------------------
-- Executing Dial("SIP/msn-6297", "H323/73952389512@peer:1720") in new
stack
Allowed Codecs:
Table:
G.729A{sw} <1>
G.729{sw} <2>
G.711-uLaw-64k <3>
G.711-ALaw-64k <4>
2005 May 24
0
H323 integrated Asterisk support
Hi all,
I used oh323 support from inaccess. It work very well.
I would like to test h323 integrated support.
This my problem when I test it :
I cannot heard any thing in both way.
The test is : SIP --> Asterisk --> H323
This is th debug trace from h.323 :
-- Executing Dial("SIP/someaccount", "H323/0033172897104@somehost") in
new stack
2005 Jul 03
0
H323 with GSM codec is not working
Hello,
I'm trying to use the GSM codec with Nufone H323 but it's not working.
Does somebody has some idea? Have I missed something?
Thanks!!
Celso Fassoni
Some additional info:
(I'm using CVS-HEAD - downloaded today)
monkey:~# cat /etc/asterisk/h323.conf
[general]
port = 1720
bindaddr = 192.168.0.100 ; this SHALL contain a single, valid
IP address for this machine
2005 Jan 27
0
Problem with OpenPhone->Asterisk
Hello all,
I just installed Asterisk with H323 support (chan_h323 from Jeremy
McNamara). But experience problem while connecting OpenPhone to Asterisk
Here is h.323 trace:
5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP
Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720,
handle=27
5:37.444 H225 Answer:9cc1250 transports.cxx(564) H225
2008 Oct 18
1
strange h323 delay issue
Hello,
I have a strange h323 issue. After executing command
"Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18
22:32:23. Meanwile I have sniffing traffic on port 1720. The call was
established just at Oct 18 22:33:03 (New H.323 Connection created.) and also
packet sniffer grabs the h323 invites at this time also. So my question is
what
2005 Jan 06
0
H.323 to SIP extension
Greetings All-
I have an * box with the NuFone H.323 channel driver installed.
I also have an Altigen VoIP system with a PRI to the PSTN.
I can sucessfully make a call from a SIP extension (snom190)
to an H.323 extension (altigen phone)
The thing I can't seem to make work is a call from a H.323 phone
to a SIP extension.
Here's the layout:
2009 May 06
0
problems in h323 channels
Hi, all!
when my h323 phone dial in Asterisk system, i can hear nothing. and
the following is the log slice i picked from /var/log/asterisk/full.
ps: i am using red hat AS5 kernel 2.6.18-53.el5,Asterisk-1.4.24.1,
pwlib_v1_11_0, openh323_v1_19_0_1.
Best
Regards!
81948 [May 6 10:07:34] VERBOSE[11579] logger.c: -- Remote UNIX connection
81949 [May 6 10:07:51] VERBOSE[29627] logger.c:
2005 Jan 13
0
Oh323 compilation errors
Hi, well, I really need your help here. I have tried compiling oh323 many
times and I always get the following error when trying to "make opt" open
h323. Any ideas?!
Compilation Error:
--------------------------
g++ -o obj_linux_x86_r/simph323 -s -L/root/pwlib/lib -L/root/openh323/lib
./obj_linux_x86_r/main.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread
-lssl -lcrypto -lexpat