search for: annexe

Hello Some questions about codecs.. What's the difference between the this codecs? Which is used by asterisk? Thanks Juan Salas

Hello, I am trying to build R-1.3.1 on hpux10.20 with gcc and HP's fort77. here is my configure: ./configure --prefix=/home/absd00t/local --with-readline --with-gnome=no --with-f77=fort77 I also set CXXFLAGS= -I/usr/include/Motif1.2 -I/usr/include/X11R6 -I/home/absd00t/local/include and LIBS= -L/home/absd00t/local/lib so that readline.h and libs would be found. I have parts of gnome libs so

On 07/02/2015 04:44 PM, David Keaton wrote: > On 07/02/2015 03:17 AM, Kuperstein, Michael M wrote: >> You want to redefine ["won't break the program"], by specifying a new >> abstract machine, which is >> more conservative than standard C/C++. The proper way to do that would, >> I believe, be to work towards setting up a working group within the >>

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

On 07/02/2015 05:43 PM, David Keaton wrote: > On 07/02/2015 05:30 PM, Philip Reames wrote: >> >> >> On 07/02/2015 04:44 PM, David Keaton wrote: >>> On 07/02/2015 03:17 AM, Kuperstein, Michael M wrote: >>>> You want to redefine ["won't break the program"], by specifying a new >>>> abstract machine, which is >>>> more

...and return the adequate formula in a main function. This is done through a script. When I type the formula in the main function, it works fine. When I attempted to use a second function (with conditional choice) it doesn't work any more. May you guide me? http://www.nabble.com/file/p18811385/annexe.R annexe.R ----- B. Boulinguiez benoit.boulinguiez at ensc-rennes.fr Ph.D. in Chemistry Ecole Nationale Sup?rieure de Chimie de Rennes 35700 Rennes (FRANCE) -- View this message in context: http://www.nabble.com/Trouble-when-I-call-a-function-within-another-tp18811385p18811385.html Sent from th...

Hi zozo, sorry to bother you; I've been trying to get a cisco 7920 phone (that's the cute wifi/wireless modell) to work with asterisk (no success so far) and ran across your chan_sccp module; is there a current cvs version that I could base my tests on? tar version 0.1 seems to be ~2003-09-12, cvs access as posted on http://theo.me.uk/pages.shtml?page=sccp doesn't seem to work any

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their

Dear I try to realize one scatter matrix which draws *one single variable to all variables* with *regression line* . You can see my eviews version in the annex . How can I draw this graph with R studio? Sincerely Engin YILMAZ

Dear R Staff My working file is in the annex. "g1.csv" I have only 2 columns. Rice and coke. I try to execute following(below) function, but do not work. Because "Coke" value has NA values. I try to add "na.rm=True" to the function but do not work How can I solve this problem with this function or another algorithm? (Note: I have normally 450 columns) Sincerely

LOL. Great reply Jim. (N.B. Jim's conclusion is "debatable" by a judicious choice of seed. e.g. set.seed(79) suggests that making the request more readable will actually lower the number of useful answers. :-)) On Mon, Nov 27, 2017 at 11:42 AM, Jim Lemon <drjimlemon at gmail.com> wrote: > Hi Engin, > Sadly, your illustration was ambushed on the way to the list. Perhaps

Hi, I have been working on developing Link Grammar interface, so as to use POS tagging while indexing the documents. The interface header as well as implementation file have been completed and you can view them at < https://github.com/sehaj-sk/xapian/commit/052d634e1986bcf5607e43f52ac3e07646920196> and <

Dear Berger and Jim Can you see my eviews example in the annex? (scattersample.jpg) Sincerely Engin 2017-11-27 13:27 GMT+03:00 Eric Berger <ericjberger at gmail.com>: > LOL. Great reply Jim. > (N.B. Jim's conclusion is "debatable" by a judicious choice of seed. e.g. > set.seed(79) suggests that making the request more readable will actually > lower the number of

Dear R Staff You can see my data.csv file in the annex. I try to count non-zero values in dataset but I need to exclude NA in this calculation My code is very long (following), How can I write this code more efficiently and shortly? ## [NA_Count] - Find NA values data.na =sapply(data[,3:ncol(data)], function(c) sum(length(which(is.na (c))))) ## [Zero] - Find zero values

...mend supporting all modes, but >>> I'm not sure I want to say MUST for that. >> >> I guess you already have my idea about this: all modes should be supported >> unless you know you won't have issue. >> >> On good thing with g729 and its extension (g729 annexe b?) is that you >> can still >> receive g729b if you support only g729: this is transparent (as far as I >> understood it). >> >> For speex, the modes are not transparent and thus, If I was the one >> to choose, I would add in the draft: ALL MODES MUST BE SUPPORT...

This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message

Has it been a whole week since a question about VAD? :-) I am curious why the VAD algorithm described in G.729 Annex B is not used by Speex. Complexity, Patent, Quality, or what? -mjc

We appear to be having a problem with our asterisk setup. We have a cisco AS5300 with pri lines coming in and passing the calls onto asterisk then too the sip phones. the phone call from the sip phones (7960's) appears to be ok nice and clear including the user who has called in. but if your the user who has called in its all crackley sounds really bad when they speak. i believe this

Hi ! On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone button (or select line with line button - which automatically put second line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5, chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of 7960 is

Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last