similar to: 3G-324M Open Source

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

Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does

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 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 Does Sangoma 8-port card A108 support PCIe version 2.0 ? The cards is here 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 with multiple A108 with PCIe on the same motherboard that supports PCIe 2.0 ? Any comments

Hi, Is there any future possibility that Asterisk will be compatible with connection to 3G video mobile phone such as Nokia 7600, Nokia 6630 and many ohters in Japan, Europe and HongKong? If this become possible, H.323 video clients and 3G mobile phone will be able to share video conversation, which will be huge in those countries. In Japan, more than 3 million 3G video mobile phones are

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

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)

Thanx for the reply. The reason i wrote echo is, i was running the script on the command line, and i wanted to see if the particular function is running. Just like i do debugging in c++. I didn't know that it sends messages to asterisk. But again i was not able to see any message on asterisk server. One thing, it must be possible to run the php script file on command line, since it was the

HI list-users, Greetings!! I have been using call file, i playback my file using * application:playback* and once the playback is over the call is disconnected. Is there any way it can wait and also record the dtmf inputs once the playback is over. Thanks in advace Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad contact at 9793905858 email:

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

Some days back there was a mention about an open source image editor that people uses for application layout designing. I have forgotten about the name of that software. Can someone send me the link? Silvy Mathews

Hi all, I think i understand the problem, actually I have two asterisk server. In the extension.conf file on one server I have added exten => 3923903,1,GOTO(s,1,3923903.conf) which reads the corresponding conf file when ever the extension no. through PSTN is called and learns the location of inbound.php which contains the IVR script to be executed. Now what i want is that through this

Thanks a lot Danny. I have done the part of playing a file by creating a context in my dialplan. Now I am puzzled as i wish to store the DTMF inputs done by the users who is listening to the playback. I found there are ways, but some specific way by which it is not stored in file but conveyed directly to the asterisk server. When the call landed up on the softphone, i pressed keys the

Hi All, I have a data frame of three columns, all of which names. The names in the three cols overlap up to a point, so I might have *Harry* in all three columns, *Tom* in cols 1 and 3 and *Bob* in col 3 only. harry paul bob anita harry tom frank jack harry tom pete ben .... I want to turn the names into numbers, BUT I want the numeric code for, say, *Harry*, to be the same on all

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 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.

Hello, I have been trying to get an ANOVA table for a linear model containing a single nested factor, two fixed factors and a covariate: carbonmean<-lm(C.Mean~ Mean.richness + Diversity + Zoop + Diversity/Phyto + Zoop*Diversity/Phyto) where, *Mean.richness* is a covariate*, Zoop* is a categorical variable (the species), *Diversity* is a categorical variable (Low or High), and

In R 3.5 and later you should not need to gc() -- that should happen automatically within the connections code. Nevertheless, I would recommend redesigning your approach to avoid hanging onto open file connections as these are a scarce resource. You can keep around your temporary files without having them open and only open/close them on access, with the close run in an on.exit or a

Actually my problem is not related to sip.conf and extensions.conf. I have used only standard files from martin pdf which are given as example. I am able to call some system connected over LAN, when each has a softphone and are registered on a asterisk server. But now what i want is instead of using the softphone I write a function in my file which will be executed when the call is placed. In that