New answers tagged asterisk
I've not tried it, and its effectively a special case of polling the logs (but made easier). Why not run "Fail2Ban" with a custom rule searching for "UNREACHABLE" and "Reachable" responses combined with a script to automatically update your database.
There is an easier way to do this now. Navigate the "Asterisk SIP Settings" page and simply tick the codec. You can also use this page to change the priority of the authorized codecs... http://pbx.yourdomain.com/admin/config.php?display=sipsettings
You're not going to get anything as rich as 3CX assistant without doing a lot of the assembly yourself (Ikea style assembly, not saw-the-tree down style). I support several asterisk installations and only a few of of them haven't presented headaches. If you have something that works and is not prohibitively expensive I wouldn't trade a few dollars for a ...
In freePBX, these are the steps: Add a system recording (Admin -> System Recordings) Create an Announcement using that recording. (Application -> Announcements) Set the "destination after ring" to whatever you want to (probably whatever extension somebody is picking up) Set the announcement as the destination on the appropriate inbound route ...
You don't have a username= set in your extension (often, just the extension number - it doesn't matter what it is, it just needs to match the client) The log sample you have shows an extension 12 trying to register, and not sending any Authorization header, which looks odd. CSipSimple works fine for me using the extention number as the username (and of ...
The simplest, lazy way around this is to set your asterisk box in a 1:1 nat config (often called DMZ host on home routers) and to make sure externip= is set in sip.conf. The usual troubles with SIP and NAT are: SIP headers contain call source and destination information (IP addresses) that may not be reachable to/from clients and servers behind nat A ...
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