similar to: Alternative to Adobe Audition 3 for G723 > G711 uLaw ? (old Cool Edit Pro)

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

Ars Technica reports of ?abandoned projects? on SourceForge being taken over by SourceForge with adware installers added to executables. arstechnica.com/information-technology/2015/05/sourceforge-grabs-gimp-for-windows-account-wraps-installer-in-bundle-pushing-adware/ This might be something to be wary of with FLAC considering the main focus for FLAC has moved from SourceForge to Xiph.org.

Hello, I just CVS'd today and now I'm getting these errors when I call one grandstream phone to another both using 711U: NOTICE[1225991360]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from ULAW to G723 NOTICE[1225991360]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from G723 to ULAW NOTICE[1225991360]: File channel.c, Line 1476

http://philosecurity.org/2009/01/12/interview-with-an-adware-author The inherent insecurity of Windows is deeply frightening. I hope we don't emulate it *too* well. - d.

Hi, I have softphone with a g723 codec, my question is how do i set it as Pass thru in Asterisk? cheers, Aby Azid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080424/b442d5af/attachment.htm

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

hi everybody, I try to record a call with *1 - one touch record feature in g723 format. exten => _00[1-9].,1,Set(TOUCH_MONITOR_FORMAT=g723) exten => _00[1-9].,n,Dial(SIP/${EXTEN}@ext-sip-account,,wW) I have chosen g723 format because in my CLI> show translation there is no translation between g723 format and wav (default for *1 feature). After pressing *1 sequence I have two

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

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 --

I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan

All, Our users have a softphone client that supports the G723 Codec which we want to use for bandwidth reasons, however we do not wish (or have the funds) to license the codec on our Asterisk servers. We have G723 pass-through working between the clients just fine, however calls fail when terminating with Asterisk itself (i.e. Voicemail) or out to the PSTN due to transcoding issues. If it

Hi: How to install and set up my asterisk server with G723 codec to send and receive calls using it. Thanks in advance; Wassim _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

Ok, My provider is sending a call to me via ULaw but Asterisk isn't picking up on this, I've only allowed ulaw, I disallow=all and then allow=ulaw in my sip.conf and that's the only thing I allow, but when my provider sends me the requests, I get an error about No Compatible Codecs: 17 headers, 8 lines Using latest request as basis request Sending to 67.19.245.213 : 5060

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

Hi, I've got OpenSuSE Linux 11 KDE 3.5.10.33.2 and kernel-pae 2.6.25.18-0.2-i.686 and Wine 1.1.13 Now I'm trying to install Adobe Audition (I tried several other programs for Linux, but they can't convince me due to limitations, that is, not those possibilities like Adobe Audition). I read so many "stories" on the internet that Adobe Audition could work thru Wine under Linux,

Has anybody found the plug-in for Adobe Audition (formerly Cool Edit) decodes just fine, but doesn't seem to compress? I've tried saving .wav files to the compressed FLAC format in Adobe Audition & they appear to be the same size as the uncompressed .wav. The FLAC frontend works perfectly for me, & I can compress through Exact Audio Copy successfully... just not in Audition.

I'm running Wine 1.1.10, configured as Windows XP, and a version of Ubuntu from four days ago. The appDB says that Adobe Audition 2.0 works fine (http://appdb.winehq.org/objectManager.php?sClass=version&iId=13422), but I haven't had any success yet. No exe file was installed, audition doesn't show up in wine's list of installed programs, and there's just generally no way to

The developers of Opus can help the adoption of Opus greatly by writing a filter to load/save files for the commercial Adobe Audition app. The older Audition 1.x/2.x/3.x versions, before Adobe went all Creative Cloud. Just a suggestion. Even if it's a GUI filter front-end for opusenc. Adobe abandoned their legacy plugins which supported a very wide range of formats - even HE-AAC, WMA Pro,

Thanks...which version of IPP did u use ? I do not have Makefile file....there is only a .sh script Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Juan Salas Inviato: luned? 3 ottobre 2005 15.41 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: