Hello list, I am glad to announce that XC-AST version 0.9.0 is out today. New functionalities include: * Though not yet available to the end user, this release inclued the basis of the Outbounds Call Manager that will be released for 1.0. If you update from a previous version, have a look at the UPDATING.txt to understand how to upgrade your database schema. * The realtime visualization now shows only calls within a given toime frame. This is becaise sometimes Asterisk does not write complete information for a call, and such" orphan calls" used to hang forever as open calls. It is now possible to configure the SLA for your site, defining the maximum observation period and the interval. * Due to popular demand, an explicit "delta" from period to period was added. * It is now possible to see waiting periods and call durations in real-time (as long as your system clock is the same or is synchronized to the Asterisk clock; in any other case you will not see relevant iformation and should turn off this feature). * Fixed a bug with the English pages where a percentage was showed with the Italian locale * Assorted minor bugfixes As always, the product is available free (as in beer) to smaller installations and * enthusiasts around the world to use and hack. Future plans include: * Outbound calls management (most of the code is already present in v0.9 but there are quite a few quirks yet) * Standalone application (so you won't need to install a servlet container just to test XC-AST) XC-AST can be downloaded at http://demo.xcept.it/xc-ast As always, any comment is appreciated. Thanks l. -- Assum est, versa et manduca.
On Sun, 2005-05-29 at 00:21 -0700, VOIP Consultant wrote:> I am in the process of making and SS7-to-VOIP implementation (basically > Astersisk/SS7 implementation). I have researched the subject, but the > results are not encouraging. There is an open SS7 initiative that is > basically stalled. Is there anyone who can post their opinions or > experience or suggestions on the matter? > > My goal is to set an * server with SS7 support to an Alcatel switch. >And some not so stalled http://www.voip-info.org/wiki-Asterisk+SS7 After several months of live testing VoIP operation in Europe and Asia, SS7 is now available for Asterisk. The LIBISUP solution is fully integrated with Asterisk and does not require any additional external equipment. The product is immediately available under the Digium commercial licence.\ For more information please write to markku.korpi@cosini.com and describe your SS7 project. Its not free but you will not find anything that is that is certified, which is often a requirement. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050528/a81c98e2/attachment.pgp
I am in the process of making and SS7-to-VOIP implementation (basically Astersisk/SS7 implementation). I have researched the subject, but the results are not encouraging. There is an open SS7 initiative that is basically stalled. Is there anyone who can post their opinions or experience or suggestions on the matter? My goal is to set an * server with SS7 support to an Alcatel switch. Thanks C. Savinovich