similar to: Sangoma A108 PCIe 2.0

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

Hello everyone, I'm considering getting me a quad-core Dell PowerEdge 860 to run Asterisk. Anyone else using this model? Any tales of woe and sorrow I should know about? Then, in a couple of weeks, I'm thinking of getting a Sangoma A108 and giving that a try. Same question with that one - any quirks I should be aware of? Girts -------------- next part -------------- An HTML

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 We need to evaluate some open source project that supports 3G-324M on top of Asterisk. What do you recommend ? What has been your experience ? Thanks. regards, Anita Hall, Simmortel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100729/b7fcd340/attachment.htm

Hi Guys, Looking for a powerful box that is compact, can take two hard drives for Raid-1 (no SSD, too expensive), have at least two Gig ports or two 10/100mbps ports. Fit two PCIe or one PCIe card plus it's daughter card which needs as much room as a PCIe and doesn't need the actual slot. That is for Sangoma A200 + Daughter card. I would be really glad if I can get some sort of a KVM

Hello, All. I am currently running a test configuration for the telephony engineer here. I have two Dell PE servers, each with 2 A108 Sangoma T1/E1 cards. Here is a rough drawing: Ameritec Call Generator 8 T1 ISDN/PRI lines ---> Asterisk ---> Asterisk ---> Ameritec Call Generator Sangoma A108 Sangoma A108 Terminating 8 ISDN

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)

I'm looking at setting up a new machine to run CentOS 5 and a few VMWare machines to test Windows XP, Vista, and 7. I'm working with a custom PC shop and they've recommended I use either the ASUS PT6 or Intel DX58SO. Any feedback good or bad on either of these? Thanks! --Chris

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

On Wed, 2018-10-17 at 10:50 +0800, Han Han wrote: > In libvirt, I found pcie-expander-bus controller doesn't support pcie-to-pci-bridge and pcie-switch-upstream-port. [...] > # virsh -k0 -K0 define /tmp/c.xml Aside: the -k and -K virsh options are documented as -k | --keepalive-interval=NUM keepalive interval in seconds, 0 for disable -K |

On 10/17/2018 08:56 AM, Andrea Bolognani wrote: > On Wed, 2018-10-17 at 10:50 +0800, Han Han wrote: >> In libvirt, I found pcie-expander-bus controller doesn't support pcie-to-pci-bridge and pcie-switch-upstream-port. > [...] >> # virsh -k0 -K0 define /tmp/c.xml > Aside: the -k and -K virsh options are documented as > > -k | --keepalive-interval=NUM >

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

I was working with the rmetrics portfolioBacktesting function and dug into the code to try to find why my formula with 113 items, i.e. A1 thru A113, was being truncated and I only get 85 items, not 113. Is it due to a string length limitation in R or is it a bug in the strsplit or gsub functions, or in my string? I'd very much appreciate any suggestions ============Input script:

In libvirt, I found pcie-expander-bus controller doesn't support pcie-to-pci-bridge and pcie-switch-upstream-port. Version: libvirt-4.9 # cat /tmp/c.xml ... <controller type='pci' index='0' model='pcie-root'/> <controller type='pci' index='1' model='pcie-expander-bus'> <model name='pxb-pcie'/>

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