similar to: Openphone implementation of Speex Codec's descriptions help

Date: Fri, 10 Sep 2004 16:37:33 +0300 > From: Michael Manousos <manousos@inaccessnetworks.com> > Subject: Re: [Asterisk-Users] Problems with 0penh323 Channel Driver > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <4141AE1D.3020403@inaccessnetworks.com> > Content-Type: text/plain; charset=us-ascii;

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

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

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

Hello- In order to satisfy a customer requirement, I've just build H.323 under asterisk (using the specified versions of OpenH323 & PWLib, and trying to follow the instructions religiously), and it seems to have come up fine. When testing with with OpenPhone (Windows version 1.8.1) and NetMeeting, I've gotten some intermittent results however. All my calls are from a PC to asterisk -

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:

What is the configuration of H323.conf and openphone in order to run openphone and asterisk together ?

I figure that I've successfully loaded and compiled the h323 module into asterisk I can successfully place a call from openphone to a sip phone (snom200) So I figure that the h323 module is working. The question I have is how do I return a call from the sip phone to openphone? I get an error message Sep 7 17:09:49 NOTICE[110992304]: chan_h323.c:861 oh323_request: Asked to get a

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

My apologies if this question has been answered previously. However, I found that it was nearly impossible to search and find since anything can cause a segmentation fault. Problem. When Asterisk is booting up the h323 modules and a client tries to connect before Asterisk/h323 is finished booting, the program seg faults out and doesn't load. I thought about putting this into the inittab,

>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

> What is the maximum frame-length that libspeex will produce for narrow, > wide and ultrawide encoding? In normal operation (no in-band side information, like requests, ack, stereo, ...), the max size for a frame is 62 bytes in narrowband, 106 bytes for wideband and 110 bytes for ultra-wideband. Jean-Marc -- Jean-Marc Valin, M.Sc.A. LABORIUS (http://www.gel.usherb.ca/laborius)

https://bugs.freedesktop.org/show_bug.cgi?id=103382 Bug ID: 103382 Summary: Flickering / Artifacts on TTY with GTX 1060 Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau Assignee:

Seems to me that they are using speex already. Ethereal shows that the voice RTP stream's payload type is 103. According to this page [ http://www.openh323.org/pipermail/openh323/2004-June/068705.html], 103 is SpeexNarrow-8k, although this email from Craig might not be correct. - Cheng > Message: 2 > Date: Thu, 25 Aug 2005 04:15:44 -0500 (CDT) > From: Ashhar Farhan

Hi, I have downloaded the latest stable version *1.0.2 opus audio codec* from the following link: http://www.opus-codec.org/downloads/ *Details of OS & CPU:* OS : Microsoft Windows XP CPU : intel core 2 Duo cpu I am using Microsoft visual C++ 2010 Express to build the codec (opus.vcxproj available in the package downloaded). I am able to build it & could generate the application

Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no

Hi, I have downloaded the latest stable version *1.0.2 opus audio codec* from the following link: http://www.opus-codec.org/downloads/ *Details of OS & CPU:* OS : Microsoft Windows XP CPU : intel core 2 Duo cpu I am using Microsoft visual C++ 2010 Express to build the codec (opus.vcxproj available in the package downloaded). I am able to build it & could generate the application

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we

> Framesize always refers to the decoded data frame size in samples. > Framesize is dependent on the encoding mode > Narrowband (8kHz): framesize = 160 samples = 320 bytes of PCM > The size of the encoded data depends on the quality setting, so if you > know for instance that you are using quality 3 on narrowband, that is > 119 bits of encoded data per frame which is rounded to