similar to: E3 Card on Asterisk ?

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience

Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of

Hi Does anyone have experience with Synway cards like SHD-240D-CT/PCI with asterisk and SynAst driver ? Are they any good ? Do they really run on Asterisk ? Thanks. Anita Hall, Simmortel Voice www.simmortel.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100913/c97492c4/attachment.htm

OK. Now I show my ignorance. What's the difference between a T-1 and a channelised T-1? I see that Covad's voip service (formerly GoBeam) requires a channelised T-1. Then I read recently on the list that many T-1s being installed are actually HDSL....which would be not a T-1 at all...right? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

Hello, I am trying to install FreeBSD 9.2-RC4 on Supermicro server with Intel Xeon E3-1240 v3 processor. The processor has 4 cores with 8 threads. However only one core is detected (with two threads): FreeBSD/SMP: Multiprocessor System Detected: 2 CPUs FreeBSD/SMP: 1 package(s) x 1 core(s) x 2 SMT threads cpu0 (BSP): APIC ID: 0 cpu1 (AP): APIC ID: 1 The motherboard is Supermicro X10SLM-F. All

I am getting a rake aborted error and I suspect that I am missing a package on my system since the app works for a friend on this computer. Here is the terminal output of the error: anita@anitas-computer:~/sandbox/shovell$ rake db:migrate (in /home/anita/sandbox/shovell) rake aborted! Could not find table ''stories'' (See full trace by running task with --trace)

Having a god-awful time installing Rails. I started with the instructions found on this page: http://articles.slicehost.com/2007/11/23/ubuntu-gutsy-mysql-and-ror I get the following error: vbfischer@ubuntu:~$ sudo gem install rails [sudo] password for vbfischer: Bulk updating Gem source index for: http://gems.rubyforge.org/ Updating metadata for 13 gems from http://gems.rubyonrails.org/

Sorry for the delay here. For some reason the mail from Joshua Colp failed to deliver to my mailbox. So, anyway, I've set up a local scenario on my computer a PJSIP client and Asterisk 11.17.1 (On a fedora linux workstation) with the settings listed below. In this scenario I've used UDP, but I want a configuration that can be used with any transport protocol. I can see that the context

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Mon, 21 Sep 2015 06:48:52 +0000 > Emil Ohlsson <emo at svep.se> wrote: >> [sip-im] >> exten _X!, 1, NoOp(Got message) >> exten _X!, n, Answer() >> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...) >> exten _X!, n, SendText(Message received) > > I am not

Hello, I have a windows application called Anita (terminal emulation sofware from www.april.se) that runs just fine on Linux with WINE. The problem is that Anita needs to open an HTML page and display a PDF file. It works on Windows - terminal emulation software displays the PDF file without any problems but it doesn't work on linux. It displays a blank page. My guess is because on Windows

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations <sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it

Hi List, I have stupid question but I want to know it. Why we use the PRI insted of BRI ? Just for the sake of number of lines or any thing else ? And why SIP is used for making calls rather then IAX? Even we know IAX takes 1 channel for making calls? ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Reader -------------- next part -------------- An HTML attachment was

Hello, I am trying to use nlm to estimate the parameters that minimize the following function: Predict<-function(M,c,z){ + v = c*M^z + return(v) + } M is a variable and c and z are parameters to be estimated. I then write the negative loglikelihood function assuming normal errors: nll<-function(M,V,c,z,s){ n<-length(Mean) logl<- -.5*n*log(2*pi) -.5*n*log(s) -

Hi, I have A101DE Sangoma Card( http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html ) lspci shows as 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card [root at asterisk ~]# lspci -vvv -s 03:04.0 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: Unknown device

what is the difference between FreeSwitch and Asterisk , whitch one is more scalable and reliable? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

Hi everybody, This is the console output of the asterisk server. debian-te410*CLI> sip set debug peer 2002 SIP Debugging Enabled for IP: 172.26.48.113:5061 I have a sofphone with user 2002 registered on the server on the ip 113. I am trying to place a call to the sofphone on this ip. I have written a simple php script which utilises the exec_dial function inbuilt in phpagi.php file. I have

Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards

Hi everybody, I've found a very interesting problem in R: the if-else statement doesn't work in a main program. Sounds crazy, but true. I tried this very easy example, and I got syntax error at the "else" line. Example: ------- a <- 1 if ( a == 1 ) print("yes") else print("no") -------- I tried on Windows and on Linux, but none of them works.