similar to: [LLVMdev] GSoC proposal: New front-end

Hello everyone, I wonder if there's a product that I can install on my debian-based server to extract CDRs (it'd be better if Excel's downloads are available), also it would be desirable if I can access additional table to update rows (e.g. sip for realtime) Please let me know what you know. Best Regards, Elder D. Arohuanca dCAP Lima - Peru -------------- next part --------------

Hi again, Thanks Ted and Marc its works. But of course after pulling in in some real life data I discoverd one hitch. Often there are missing intervals. For example: from <- c(0, 1.2, 4.0, 4.2, 5.0, 25.0, 30.1, 45) to <- c(1.2, 4.0, 4.2, 5.0, 25, 30.1, 36.2, 50) intensity <- c(0, 1, 3, 2, 1, 0, 2, 5) barplot(intensity, width = -(to - from), space = 0, horiz = TRUE, ylim =

I have a existencial problem. There are some provider that offer the service of bandwith asimetric as download/upload link as for example 512/256. but most of them offer not exclusive this amount of transmission or reception capacity. They usually offer the service with more users as a ratio of 1/10. How I can design a appropriate diagrama with htb where the amount of bandwith could vary.?

Hello all, I need the bootrom.ld file to set up some Polycoms I have Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A I've publiched on my FTP files downloaded from http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html (3.2.3 combined and split zips) but my phones are still showing the message: "error, application is

Hello everyone, I've suffering cut offs after 6 or 7 seconds a call is answered, incoming calls are working fine, but outgoing ones show the gollowing messages when are being dropped: [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical Response) -- See

Hello Friends: I've just installed Asterisk 11 on my Linux (debian) server but it is not starting up when trying with "asterisk -vvvvvvvvvvc" and "service asterisk start". Starting process just stop and shows: "Illegal instruction" as final output. Looking at logs I fouind at /var/log/asterisk/messages : [Nov 25 11:09:26] Asterisk 11.6.0 built by root @

Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648 Peer User/ANR Call ID Seq (Tx/Rx) Format Hold

Hello guys, I have this problem when a call is received in my PBX: (Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) --> (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for many seconds until it hangs up. The problem is that Telephone Company is billing me

Awesome, thanks! As per your suggestion, I read the description of these two projects: Advanced Heuristics for Ordering Compiler Optimization Passes Machine learning and compiler optimizations: using inter-procedural analysis to select optimizations and they are amazing! Indeed, they are very close to my interest in autotuning. I didn't see them on the list before. If I choose to focus on

Hello everybody. Last monday I sent an email to the LLVM dev mailing list saying that I was looking forward to working on these GSoC projects: *Advanced Heuristics for Ordering Compiler Optimization Passes* and *Machine learning and compiler optimizations: using inter-procedural analysis to select optimization* I currently do an undergraduate research on compiler autotuning of Rust code, more

hi every body. Im new in this program. Im traying to install R in linux suse10.0 in two following form: a) with the file R-2.5.0.tar.gz b) and the rpm file : R-base-2.5.0-2.1.i586.rpm ****** In the first case a) when i uncompressed and type: linux:/opt/R/R-2.5.0 # ./configure the followind message is showed linux:/opt/R/R-2.5.0 # ./configure checking build system type...

Hi List, I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was easy and straightforward with Debian 6.0.6, but when I introduce this command on CentOS: #./ast_tls_cert -C 10.200.108.17 -O "MyCompany" -d /etc/asterisk/keys/ I got this error message: hostname: Unknown host Same result happens when using server's hostname: #./ast_tls_cert -C ast-centos -O

Dear list: When I call an specific number on the PSTN, the provider who holds the destination number give back an specified sound just after admitting their incoming calls. Is there a way to allow Asterisk to compare sounds received to decide what is the Telco answering the call? I'm planning to do it to select the right provider to route further calls at least cost. In my country there are

Hello friends, I have been experienced suddenly stops for my Asterisk server, I do not why is it happening. Asterisk's debug messages only tell me I have lacked g729 codec for translation to one peer minutes before the crashes occur [2014-05-27 09:48:30] WARNING[15384][C-0000017c] channel.c: Unable to find a codec translation path from (ulaw) to (g729) [2014-05-27 09:48:30]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am a newcomer to CentOS and been lurking the list for a week or so. This morning I installed CentOS 4.1 on a Compaq Evo (Pentium IV, 512MB RAM) After the initial boot just after installing, the console if constantly filled with "eth1: mismatched read page pointers 0 vs ff". This message appears at least every 40 seconds. The same

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, About a week ago I was forced to migrate my main (production) email server from RH9 to Centos4.1. The installation has dovecot-0.99.11 + postfix-2.1.5 and >600 mbox accessed email accounts. Number 1. Some users accessing thru squirremail (installed in my webserver) can read their INBOX at /var/spool/mail directory but when trying to

Hi everybody, I'm very glad to meet you. Congratulations to Samba staff, for them open source software is well known as high quality product. I've got a problem now, since I changed of server. First of all, my old server HP ML330 G3 suffered a little problem, and was replaced by an HP ML150 G2. The older server had RedHat 9 installed with Samba 3.22. This server runs Fedora Core 5 and

Hi, I am developing a VoIP client, and I want to use speex as the encoder. I'm using python as the programming language for the project. However, when I try to decode frames recorded from the microphone and then play them, I got an error, which says that I should give a string as an argument in order to play, not a list (speex.decode returns a list) Any idea how to fix this?? Greetings from

Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif Unsuccessful Asterisk Command: same => n,System(mutt -s "New fax" elder.arohuanca at

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: