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Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other

...uy a ready built pc from ebay but now i think i will built a 19" rack case so i will built in a 1 - 1,5ghz intel pentium with 256MB ram. And when my Asterisk is runnig i will setup another at my office and i will connect the two asterisks. Do you think the cpu is big enough? Do you think i can hadle up to 3 simultanus calls on sip with my internet connection? Moritz ----- Original Message ----- From: "William Suffill" <william.suffill@gmail.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, August 07, 2004 8:23 PM Subject: Re: [Asterisk-Users] System Requirem...

Hi there, me again! I´m looking for a solution for the following: I have two (a bunch really..) of services that work together and have to exchange some configinformation between each other. I can get this to work in only one of three cases: 1. Works # class serviceB { file {"serviceBconfig": content => $serviceA::serviceAvar } } # # class serviceA { $serviceAvar

...as service during installation, but I'm suspecting, that this dll/exe isn't executed in such a manner, that autoroute.exe can access it when it runs. Also, when you active the GPS function in AutoRoute, wine reports, that the ports (com, lpt) get initialized, but nothing else (ie. no file hadles, no baud-rates). And then, after a while, the application reports "no gps receiver found", as if it never saw anything on the serial ports. BTW: gpsd / gpsdriver or minicom all tell me, that the gps receiver is working ok... Perhaps someone here can help me debugging this stupid progg...

...an APACHE server in the other office... will I have enough bandwidth to handle calls? Thanks so much for any insights... with a new business, a 3 year-old, a 1 year old and a very pregnant wife - I don't have a lot of time to experiment with new ideas :) So I'm trying to really get a hadle on everything before I decide to invest a lot of time trying to make this work... thanks again... dave redmore

Hi. When can we expect the next beta version of Vorbis encoder (with some tagging & encoding options I hope)? Also, today I tested MPEG+ encoder which you can download from http://www.stud.uni-hannover.de/user/73884/mppenc.zip (win encoder) http://www.stud.uni-hannover.de/user/73884/in_mpp.zip (winamp plug-in) http://www.stud.uni-hannover.de/user/73884/mppdec.zip (win decoder) It sounds

Greetings list- I'd like the ability to hangup all calls for a particular extension from the system CLI. I understand this can probably be scripted using the AMI but I'm not familiar on how to do it. Help! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105

Hi, I'm currently looking to run Asterisk in the office to replace an old PBX and would appreciate a little help. We are moving offices and will have 8 digital lines. My questions are: As there are 8 digital lines is this known as PRI? Which Digium card would be the best fit? Would you recommend looking at the echo cancellation cards? We are UK based: is caller id supported by Asterisk

In environment.rb file, I initialize a connection to some data vending servers, through TCP sockets. The connection object is global and hence the code: #environment.rb $generic_connection = ConnectionClass.instance (singleton class) $generic_connection.connect_me( this call will make the connection) The above approach is to make sure that, only one connection is made to the data vending