In FreePBX, when I select voicemail for the extension, and if the caller sent for the voicemail, and he leaved (or did not leave) a voice message, and did not press #, so the channel will stay open and this is not good specially if the call was coming from outside via the analoge lines (because the caller might hangup and the dahdi does not detect the hangup, so the channel will stay openned).
How to let the voicemail hangup automatically after waiting for certain seconds (for example after 30 or 40 second), then to hangup or jump for the next…
If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal).
Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I have to write the configuration in my hand and it will not be autogeneration, correct?
In this case, the Phone will not have any of the features that I am going to add…
I am new. Here is the code that I am playing with on CentOS 6.x
When I dial the number that corresponds w/ my SIP account I get a recording: "reached a non-working number........"
I built Asterisk a few times last year and am now back working on a similar project. In my view, there is something wrong in sip.conf I don't remember using a file that long to get a basic call set up. The format was provided to me by voipvoip.com (the SIP provider).
Does anyone have any comments please? I just want a very simple config to get my machine…
If a single voicemail account is manipulated by two parties simultaneously, a condition can occur where memory is freed twice causing a crash. Management of the memory in question has been reworked so that double frees and out of bounds array access do not occur. Upgrade to the latest release. Affected Versions
- Product Release Series
- Asterisk Open Source 1.8.x 1.8.11 and newer
- Asterisk Open Source 10.x 10.3 and newer
- Certified Asterisk 1.8.11-certx All versions
- Asterisk Digiumphones 10.x.x-digiumphones All versions
- Product Release
- Asterisk Open Source 18.104.22.168, 10.5.2
- Certified Asterisk 1.8.11-cert4
- Asterisk Digiumphones 10.5.2-digiumphones
Asterisk Project Security Advisory - AST-2012-010
Product Asterisk Summary Possible resource leak on uncompleted re-invite transactions Nature of Advisory Denial of Service Susceptibility Remote authenticated sessions Severity Minor Exploits Known No Reported On June 13, 2012 Reported By Steve Davies Posted On July 5, 2012 Last Updated On July 5, 2012 Advisory Contact Terry Wilson
Description If Asterisk sends a re-invite and an endpoint responds to the re-invite with a provisional response but never sends a final response, then the SIP dialog structure is never freed and the RTP ports for…