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CentOS Errata and Bugfix Advisory 2015:0670 Upstream details at : https://rhn.redhat.com/errata/RHBA-2015-0670.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) i386: 9834c47c99d57410a736ca2df5589999f5019e8621ab94f011b12396a8badd9c environment-modules-3.2.10-2.el6.i686...

CentOS Errata and Bugfix Advisory 2014:0670 Upstream details at : https://rhn.redhat.com/errata/RHBA-2014-0670.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 5543ae6fca2208bec2ecdb67eaf76b1f3ade9afff493821bd285f3a8b5a21b41 spice-server-0.12.4-6.el6_5.2.x86_...

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work

...to compile with MeetMe() application support enabled, nor does there appear to be a module I am failing to load that would contain this application. Is there something really obvious I am missing? Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

...n your messages for some monthly price. > > Is it interested for you? > > -- > Thanks, > Pavel Ismailov > skype: pavel.ismailov > www.flashphoner.com > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u

...p' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When replying, please edit your Subject line so it is more specific than "Re: Contents of CentOS-announce digest..." Today's Topics: 1. CEBA-2014:0670 CentOS 6 spice-server Update (Johnny Hughes) 2. CEBA-2014:0682 CentOS 6 libvirt Update (Johnny Hughes) 3. CEEA-2014:0681 CentOS 6 man-pages-ja FASTTRACK Update (Johnny Hughes) 4. CEBA-2014:0683 CentOS 6 nmap FASTTRACK Update (Johnny Hughes) --------------------------------------...

Hello, I have a number centos3.8 installations. Recently I have figured out that version of coreutils is : [ivan at algol ivan]$ rpm -q coreutils coreutils-4.5.3-28.1 while RedHat has released an bux fix errata to that package: https://rhn.redhat.com/errata/RHBA-2006-0670.html Actually I discovered that when I encounter bug that should be fixed by this update: http://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=200327 Why this package is not included in recent centos3 updates? Thank you, Ivan

So, is there anyone out there that provides rather generic but comprehensive CNAM-style directory services via SIP, to end-users? So I can put names to my calling numbers? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

I have a customer installation with an Adtran DS3 mux. The DS1's go into my Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and they tell me that they routinely drop the DS3 service to redundant back-up's and that this is a common practice that happens thousands of times to DS3 lines daily across the US without any service interruptions. They say that the

...24 alt 2 hd 127 sec 127> /pci at 8,600000/SUNW,qlc at 2/fp at 0,0/ssd at w216000c0ff87a7a9,0 3. c6t42d0 <SUN-StorEdge3510-327P cyl 35424 alt 2 hd 127 sec 127> /pci at 8,600000/SUNW,qlc at 2/fp at 0,0/ssd at w226000c0ffa7a7a9,0 4. c6t118d0 <SUN-LCSM100_F-0670 cyl 65533 alt 2 hd 128 sec 170> /pci at 8,600000/SUNW,qlc at 2/fp at 0,0/ssd at w202500a0b85a4cc8,0 5. c6t118d31 <SUN-UniversalXport-0670 cyl 8 alt 2 hd 64 sec 64> /pci at 8,600000/SUNW,qlc at 2/fp at 0,0/ssd at w202500a0b85a4cc8,1f 6. c6t119d0 <SUN-LCS...

http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/ FUD? Interesting? Boring? New news? Old news? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

We are currently seeking a Statistical Analyst to work for a large Pharmaceutical company in the US. This position is a Work From Home opportunity. If you meet the requirements below please send your resume to Christina.Kinchen at InventivHealth.com or call 407-548-0670. REQUIRED: * The ideal candidate will have strong R programming skills and 6 months, or more, work placement/intern/post-graduate work in an epidemiological research environment (regulatory, pharmaceutical or CRO) * Experience working in data mining or working with big data *...

We are currently seeking a Lead Statistician to work for a large Pharmaceutical company in the US. This position is a Work From Home opportunity. If you meet the requirements below please send your resume to Christina.Kinchen at InventivHealth.com or call 407-548-0670. A brief description is as follows: Title of Position: Telecommuting LEAD STATISTICIAN REQUIRED: . The ideal candidate will have experience with R and 1-2 years, or more, work placement/post-graduate work in an epidemiological research environment (regulatory, pharmaceutical or CRO) . Experien...

hello all, i have 2 asterisk boxes on that 1 have public IP Address and another is only have local IP address now on public IP there are some 7 DID forwarded , now i want to forward 3 DID out of 7 DID to local machine we called server B , I know there are DIal , and Switch statement in asterisk , but is there any other convenient way to do this. because if call ratio is high then my call legs

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

...r are grossly in need of correction. So, before I link it off the voip-wiki I am extremely eager to solicit the input of the community. If you get a chance and take a look, I would appreciate it. Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: