I am using the AMI interface to start calls.At one point I have a 10 second delay Wait(10) in the dialplan… During this time it seems that if I then connect with the manager interface and place a call that nothing happens till the 10 seconds is done…
Dear Forum,I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk crashed while executing meetme kick all CLI command from manager interface. The link says the issue has been closed however I am not able to identify in which rele..
I see the following paragraph in the Asterisk trunk LICENSE file:In addition, Asterisk implements two management/control protocols: the Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface(AGI). It is our belief that applications us..
A user of the Asterisk Manager Interface can bypass a security check and execute shell commands when they lack permission to do so. Under normal conditions, a user should only be able to run shell commands if that user has System class authorizati..
while testing asterisk 1.8.10-rc2 I stumbled across a weird behavior. I want to notify a snom phone to reload its configuration. For this to happen, I use the NOTIFY mechanism. I started the notify via AMI command. Asterisk is bound to udp 25060, beca..
Happy New Year to all! Asterisk 1.8.x I have a queue to which I add agent channels like SIP/300 dynamically using the manager interface. Once logged in, there SIP/300 of course rings when a call is distributed to them. How can I also get the agents c..
Thanks, looks like: sipp.sourceforge.net supports scripting… Are there sample University Curricula for teaching VOIP with Asterisk or FreeSwitch? On Sun, Oct 16, 2011 at 5:26 AM, Daniel Trybawrote: > On Sat, Oct 15, 2011 at 08:12:33PM +1100, Alec Tay..
On Sat, Oct 15, 2011 at 08:12:33PM +1100, Alec Taylor wrote: > If asterisk or freeswitch would be taught in a classroom environment, > is there someway to emulate and script emulation of users calling > in/receiving calls, transferring calls &etc? ..
Hello! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and ..
Friends, While working with the manager interface, I noticed that an originate action to a non-existing extension had a strange behaviour. Instead of generating an error, which would happen in most VoIP channels and Dahdi, Asterisk started looking ..