similar to: C&W ISDN110

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

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

Right, I'm baffled. I have: exten => s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2) exten => s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\ (caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\ VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12)) exten => s,n,MYSQL(Query RESULT1 ${DB1} SELECT\

YOu have to look for which DAG fragments are not being matched. Then you can create patterns for those using alternate sequences. When you disablea given instruction, don't just disable it, but also look at what pattern it was matching. Then create a patten for that using remaining instructions or in some cases you might have to call a library function, as in the case of floating

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 all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

Thanks for your reply. We are trying to implement a simple Mips-based CPU with just for teaching purpose, so we delete some instructions which are not commonly used, thus the task won't be too hard for students. I am responsible for modifying the compiler so that the compiler won't emit unsupported instructions. In order to avoid "can not select" error, I am trying to expand

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:

why do you want to "ban" certain instructions? is this for some architectural variant? the compiler is trying to match patterns from the target independent part of the code generator. if you remove instructions, the compiler in many cases will no longer be able to match certain patterns and you will get thos "can not select" messages. On 07/20/2012 03:05 AM, Geraint Yang

Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the conversation is perfectly normal... just not the recordings that are produced, or any spying that's going on

Hi Guys, I have a strong feeling the loads on my servers will be shooting up soon... anyone got any idea how many calls i can expect to put through a DL360: Dual Quad Core 2.33ghz 4gb RAM with 1gb allocated for a ramdisk (call recordings) This server is recording calls (mixmonitor), codec is gsm (no conversion). I know there's a lot of other things to consider like AGI scripts and such

We are encountering some problems compiling a working version of Samba 3.4.1 with ADS support on Solaris 9. We've run autogen.sh and configure successfully (config.log attached), followed by make and make install. However, despite having openldap and Heimdal Kerberos (1.2.1) installed, and including the configure switches --with-ads --with-ldap --with-krb5=/usr/heimdal, when we try a net join

Just looking for some ideas here... Single office with 1.4.26.2 - Frontend & 1.4.26.2 w/sangoma A108 Gateway I have been getting a few complaints about "caller cant hear me" or "I cant hear the caller" I've listened to the recordings and can verify what they are complaining about, with this being said, most calls are fine. I know there are alot of issues that can

Classification: NOT PROTECTIVELY MARKED Solaris 9 Samba 3.4.5 I know this isn't the sort of query that gets much response but I'd be really grateful of any advice people can offer. I'm getting really fed up with Samba as I've never been able to make it work properly. Either I'm missing something basic (probably) or it just doesn't behave in the way I think it should!

I've been trying to compile Samba 3.4.0 on a Solaris 9 server. However, when I run configure, it is only creating a Makefile-noincludes, not the standard Makefile. Looking at the config.log file, there are lines that suggest that it can't find the libiconv libraries. These are installed (version 1.11) in /usr/local so I tried re-running configure with the option --with-libiconv=/usr/local.

Hi everyone, I'm trying to get calls to record with the following setup: Using phpagi originate is called from a web application: $asm->originate("Local/" . $row['extension'] . "@sip-standard", $row['phone_number'], "sip-standard", "1", "", "", "7000"); The agent being called is extension Local/101 at

Is there a way to specify which IP address to originate calls from in a peer on sip.conf? I need to send calls from 10.1.3.10 which is a routed network through openvpn, but it's using 10.39.0.10 which is a vpn IP address - the asterisk box is the same box as the vpn bridge for the 10.1.3.0/24 network. I can't set the host as 10.39.x.x as it is dynamic. i can't change bindaddr since i

Dear all, I'm trying to replicate the panel function behaviour that Deepayan and others discuss here: https://stat.ethz.ch/pipermail/r-help/2007-April/130779.html and Oscar Lamigueiro refers to here and provides sample code for: http://www.r-bloggers.com/confidence-bands-with-lattice-and-r/ where a polygon is generated around lines on panels to display the confidence interval bounds for

Classification: NOT PROTECTIVELY MARKED Samba 3.4.5 Solaris 9 Windows 2000 AD domain Heimdal Kerberos 1.3.1 Samba is configured and the server is joined to the domain. wbinfo works as it should do, and so did getent when I had enumeration turned on. I can view and change security properties from a Windows client (as a member of the owner group). I've created a share and set permissions