Dears;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 ..
All;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 defaultcontext (from-internal).Normal..
I am new.Here is the code that I am playing with on CentOS 6.xWhen 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 simi..
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 acc..
Asterisk Project Security Advisory – AST-2012-010 ProductAsterisk SummaryPossible resource leak on uncompleted re-invitetransactionsNature of AdvisoryDenial of ServiceSusceptibilityRemote authenticated sessions Severity MinorExploits KnownNo Repor..
The Asterisk Development Team has announced security releases for CertifiedAsterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are released as versions 1.8.11-cert4, 126.96.36.199, 10.5.2, and 10.5.2-digiumphones.These releases are availa..
All,Its small issue but making a big problem for my application. I have CentOSrelease 5.8 (Final) with asterisk 1.4.41 installed. I am using 1.4.41because Flite work in this version.problem is that when I make changes on .call file to make it fut..
Im curious about the availability of Multi Function Printers with the following feature :- user feeds paper sheets in- user dials a phone number (0123456, for instance) then a hits single button- the result is that the paper sheets are scanned int..
Now and then, Im facing environments in which it could be helpful to integrate building intercom systems with Asterisk.Those intercom systems are made of :- a main panel, showing company names and equiped with a speaker, a microphone and an optio..
dearplease Help. I am continously getting this message after sip set debug on. and not getting clear voice from both side.SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received188.8.131.52;rport93Fr..