similar to: Random Zap Channel Drops to SIP

Hi all, First time poster and a relatively new R user, I'm beginning analysis for my masters degree. I'm doing a bit of work on octopus behaviour, and while it's been fascinating, the stats behind it are a bit beyond my grasp at the moment. I was hoping that somebody with more experience my be able to look at my example and offer their wisdom, much to my appreciation :-) At the most

Hi [MacOsX 10.4.6; R-2.2.1] I have a bundle that comprises three packages. I want to run R CMD check on each one individually, as it takes a long time to run on all three. I am having problems. The bundle as a whole passes R CMD check, but fails when I cd to the bundle directory and run R CMD check on a package directory. The whole bundle passes: octopus:~/scratch% R CMD check

Does anybody know if there is some tool to convert from WHIRL to LLVM? maybe some project under development? a similar project? Thanks > > --- Duraid Madina <duraid at octopus.com.au> wrote: > > Date: Fri, 18 Mar 2005 12:45:54 +0900 > > From: Duraid Madina <duraid at octopus.com.au> > > To: ahs3 at fc.hp.com, LLVM Developers Mailing List <llvmdev at

On 19/09/2007, Daniel Aleksandersen <aleksandersen+xiphlists@runbox.com> wrote: > On Wednesday 19. September 2007 19:02:06 Ian Malone wrote: > > Daniel Aleksandersen wrote: > > > Attached is a much improved version of yesterday's draft. Introducing > > > the audio:collection:artwork element to deal with album cover graphics > > > and such. > >

Hi, I'm currently building an asterisk system which should work as gateway between SIP phones and ISDN. Most parts are working very fine, but one problem occurs and I am not able to solve or debug it. Telephony from ISDN to SIP (a Sipura Hardphone) is working very well, but if the SIP Phone initiates the call, the ISDN phone rings, and a connection can be established. But no one of the two

Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make config. I think I have compiled and loaded Zaptel successfully as

Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a lot of non linux guys to make their first steps in the asterisk world Crossposted to -users. Zoa Luigi Rizzo wrote: > As a

Hi, I have asked this question months ago - i have "toggled down" all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like "A"

Hello everyone, My setup: Debian squeeze Asterisk 1.8, DAHDI, libpri, compiled from source TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800 I'm trying to get MWI for Voicemail working. In the same server I have also got an Eicon DIVA PRI card for testing purposes (it is integrated via CAPI and the chan-capi channel driver into my Asterisk). MWI works just fine there. I

Building on the great work that James Knight did on migrate-downstream-fork.py (Thanks, James!) [1], I've created a simple tool to take migrated downstream fork branches and zip them into a single history given a history containing submodule updates of subprojects [2]. With migrate-downstream-fork.py, one is left with a set of unrelated histories, one per subproject: llvm

Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting

Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron

Hello list, I wonder if anyone might be able to help me troubleshoot an attempt at porting some simple Python code to R. The function below is supposed to take a matrix containing item ratings from various users and, given a vector containing at least 1 rating and 1 missing value, employ a 'weighted slope one' algorithm to predict the missing values. The algorithm itself is fairly

Hi, I'm testing attended transfer with 3 SIP phones. I noticed about 10% of my transfers make the call drop and I get this on my log: Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: Failed to write frame -- Playing 'beep' (language 'it') Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: Failed to play transfer sound! Moreover, every

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

Dear forum, I have a install of asterisk using AMP. I followed the install guide off the AMP site. http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf When I start using amportal start or asterisk -ccccv I received this in my log. The last line is that ogg failed. I have found nothing on the web about this, and I am not even sure where to start troubleshooting. Any help

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

Hello all, as good? It would like to make a question, asterisk supports the protocol qsig, for interconnections in ISDN with equipment Siemens HiPath 4000 or same Ericsson MD110, so that it could identify to the name and the number of hosts and also to use some features of asterisk in the Siemens/Ericsson equipment. Greetings Josu? -------------- next part -------------- An HTML attachment was

Hi all, These days I installed xen 4.3 unstable from source and recompiled qemu upstream with spice support. After xl create , the windows domU started successfully and spice client can visit the VDI. However, it display blue screen before entering the windows welcome screen. Is it a bug in current upstream qemu support? My domU configuration file is: builder =