similar to: Re: dovecot Digest, Vol 36, Issue 72

Hi, I am running speex wideband for speech. While doing that I am getting Hissing sound during silence. If you want I can give you input and output files. Thanks, Shridhar -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of speex-dev-request@xiph.org Sent: Tuesday, October 09, 2007 12:30 AM To: speex-dev@xiph.org Subject: Speex-dev

Je suis absent du 2/04/2007 au 11/04/2007. Je r?pondrai ? votre message d?s mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou C?dric Buzay.

Viswanath Puttagunta wrote: > What should we do for power-of-2? I really want to avoid putting > runtime checks if nfft is power of 2 in opus_fft_float_neon. Given the tests that had to be disabled for NE10, I suspect we will not really be able to use it for CUSTOM_MODES, which should be the only time nfft is a power of 2. So I'd suggest just disabling the support when CUSTOM_MODES

thanks for your reply but i think ${BRIDGEPEER} will work only when both channels are connected. i want to get channel-id before dialing so that i can dial using that channel id. > ${BRIDGEPEER} is probably a good way to do what you want.. if Channel > A calls Channel B, and you want Channel A to "get" the channelID of > Channel B, as long as the two channels are bridged,

Martin Leese wrote: > Subject: [opus] MIME Types and File Extensions > To: opus at xiph.org > Message-ID: > <CAAzqGd_uzR646Nsdt=O2HDxLOYE2=K=5n9UOHLr3Y4BGzdVasw at mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > Hi All, > > On the Xiph Wiki page at: > https://wiki.xiph.org/MIME_Types_and_File_Extensions ... > Could somebody more

Hello All, I am currently working on a project which requires me to generate a .bc file for given .c file and open the .bc file to identify various functions and the caller callee relationship amongst them. The end goal is to generate a type of callgraph for all the functions present in the original C code. I am quite new to llvm and will really appreciate if I can be provided some pointers. I am

yes, actually this scenario is on remote servers. like SIP/XYZ at 119.18.230.20:5060 SIP/XYZ at 202.68.0.90:5678 audio is ok when dialing without using ip & port as below SIP/XYZ but when i dial using below dialstring SIP/XYZ at 202.68.0.90:5678 or SIP/XYZ at 119.18.230.20:5060 then the problem arises hope you got the idea.. Nasir

thanks for your reply but i did not meant that. ${CALLERID(DNID)} will return then number which i don't want. what i want is channel-id like if we have a user named "nasir", then we dial it as follows Dial(SIP/nasir) but actual channel-id that asterisk uses is something like " nasir-2b487e9". and on the asterisk cli we can check this when call is answered or hangup,

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Remco Barendse <asterisk@barendse.to> wrote: >> I'm not running trixbox but normal Centos 4 with asterisk installed. I tried to find some further info on this but couldn't find any. Do audio problems occur with normal Centos and the latest kernel version too? (In other words, should every centos user downgrade??) << I can't give you a definitive answer, but can

> Date: Mon, 2 Feb 2009 12:56:15 +0100 > From: friedrich.leisch at stat.uni-muenchen.de > Subject: Re: [R] Problems in Recommending R > To: thomas.petzoldt at tu-dresden.de > Cc: "r-help at r-project.org" <r-help at r-project.org>, > useR-2009 at r-project.org, paul at stat.auckland.ac.nz > Message-ID: <18822.57183.637787.426445 at

On 3 February 2015 at 01:31, Phil Wang <Phil.Wang at arm.com> wrote: > Hi all, > > I have already added support for scaled forward non-power-of-2 floating-point FFT: > https://github.com/projectNe10/Ne10/commit/79c3d787302f8d74b9bcfe6545d487cdf1b101d9 > > Two flags are added to cfg structure: is_forward_scaled and is_backward_scaled. > By setting is_forward_scaled to

Le 23/06/2011 14:00, nut-upsuser-request at lists.alioth.debian.org a ?crit : > Send Nut-upsuser mailing list submissions to > nut-upsuser at lists.alioth.debian.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.alioth.debian.org/cgi-bin/mailman/listinfo/nut-upsuser > or, via email, send a message with subject or body 'help' to >

sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows SIP/XYZ at 119.68.0.90:5060 SIP/XYZ at 202.16.34.10:5678 so dial command with unique-id i want to use will be Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT) and not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)

samba-request at lists.samba.org wrote: > Subject: > [Samba] DNS error when join domain (Win 7 -> SAMBA 4) > From: > petro at iei.org.br > Date: > 11/13/2013 07:36 PM > > To: > <samba at lists.samba.org> > > > > > I stood up a samba 4 (4.0.10) Active Directory domain controller on > a Debian Wheezy server, configured in accordance with the

Michael, rjags has updated. Thank you. BTW, I don't really know why I addressed the email to Dirk... Sebastien. Le 03/11/2015 12:00, r-sig-debian-request at r-project.org a ?crit : > Send R-SIG-Debian mailing list submissions to > r-sig-debian at r-project.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://stat.ethz.ch/mailman/listinfo/r-sig-debian

If the left and right channels are processed separately as you do, the parameter Mic-Number in speex_echo_state_init_mc should be 1, not 2 On Wed, Feb 10, 2010 at 4:00 AM, <speex-dev-request at xiph.org> wrote: > Send Speex-dev mailing list submissions to > speex-dev at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

----------------------------------------------------------------- Doesnt all this belong somewhere else besides the security lists since this isnt a security issue. ----------------------------------------------------------------- On Tue, 10 Aug 2004 freebsd-security-request@freebsd.org wrote: > Send freebsd-security mailing list submissions to > freebsd-security@freebsd.org > > To

*I'm trying to build an application that provides statistics of calls*>* and call recording. Someone told me this could be done out of band*>* with a SPAN (?) port that would replicate SIP and media packets to a*>* separate NIC without having to actually pass the real-calls thru*>* asterisk. It was explained that this SPAN port would in the SBC*>* would replicate data

Send CentOS-announce mailing list submissions to centos-announce at centos.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.centos.org/mailman/listinfo/centos-announce or, via email, send a message with subject or body 'help' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When