Re: [asterisk-users] Internal timing under load is critical ?
On 07/31/2014 09:51 AM, Chris Bagnall wrote: On 30/7/14 10:08 am, babak wrote: I am evaluating some voice broadcasting solutions based on Asterisks for more than 1000 simultaneous calls. As a matter of curiosity, what do people use these voice broadcasting solutions for? I'm genuinely struggling to think of (legal) reasons why you'd want to broadcast 1000+ simultaneous calls. Perhaps I'm just not being imaginative enough... :-) Kind regards, Chris Community and emergency notification systems are the first uses to come to mind. That said, Jane from Card Services is probably much more common. -- Daniel Taylor VP OperationsVocal Laboratories, Inc. dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk call forward for T1 incoming calls
On 04/28/2014 01:37 PM, Don Kelly wrote: On Fri, Apr 25, 2014 at 9:19 AM, Al lists asteris...@gmail.com wrote: Is there a way to divert incoming calls on DAHDI T1 channels so telco gets the diversion and send the call to new number and releasing the channel? Rusty Newton: I'm no PRI expert, but I do remember from my time working with the T1 interface cards that two B-channel transfers did something like this. Digium has documentation on that here: http://kb.digium.com/articles/Configuration/Two-B-Channel-Transfers If that doesn't help, and you have a Digium card; you might call Digium tech support to ask about it. Don Kelly adds: TBCT doesn't actually divert, but it will transfer a call. The difference, which may not matter to you, is that one of the channels must be answered by Asterisk (unless something has changed). A call comes in, you decide you want to divert it--you place an outbound call on another channel. You must either answer the incoming call or wait for the outgoing call to answer before completing the TBCT. Another quirk to be aware of: You may wish to present the caller's number to the called party as caller ID. Some carriers will not permit this. Also, the op referred to T1 channels. TBCT works on PRI, not simple T1. --Don Additionally, most providers charge for TBCT provisioning, so you would want to make certain that it is the correct way to get the functionality you are after. -- Daniel Taylor VP OperationsVocal Laboratories, Inc. dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6
I don't know what platform you are on, but if you are on Linux (and possibly BSD) you could use fail2ban to block them at the network interface. On 04/04/2014 09:00 AM, motty cruz wrote: Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- Daniel Taylor VP OperationsVocal Laboratories, Inc. dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IOPS required by Asterisk for Call Recording
On 01/24/2014 11:26 PM, Amit wrote: Thanks for response. How do I derive the requirement? I need to size IO system to record multiple calls concurrently. I ran test with following configuration Quad Core Xeon with 4GB RAM 250GB SATA disk (No RAID) Linux (CentOS 5.9) Asterisk 1.8.20 I'd suggest testing your system while monitoring with top and iotop (which should be a yum install away). That should show you your bottlenecks. It looks to me like Asterisk doesn't do compression until the call is ended, so recording to a compressed format would actually increase IO load (write, read and compress, write compressed data). -- Daniel Taylor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LUA
On 07/18/2013 08:56 AM, jacob.e.mi...@l-3com.com wrote: I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site, and I have installed via the make linux install command. I can execute lua scripts via the command line, but asterisk configure script is unable to find the installation of Lua. I am on a closed network, so no access to the internet so I am not able to just install Lua using yum. OS CentOS 6.4 Asterisk version 1.8.13.0 11.4 $ find / -name **lua** /usr/local/include/lua.h /usr/local/include/lua.hpp /usr/local/include/lualib.h /usr/local/include/luaconf.h /usr/local/lib/lua /usr/local/lib/liblua.a /usr/local/bin/luac /usr/local/bin/lua /usr/lib64/liblua-5.1.so /usr/bin/luac /usr/bin/lua You don't mention it here, so I have to ask if you tried using --with-lua=/usr/local as an argument to configure. -- Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
