search for: idokan

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

Hello, I have the following settings for manager on two Asterisk 1.2.24 (that have installed over a year ago): [user] secret = password deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 write = call,command On one server, Asterisk only react as you would expect -> sending a command without having any verbose on anything. On the other machine, I have verbose like I enabled everything,

Hello, I have a weird case, when some numbers dialed using a PRI, have an early media sounds instead of normal ringing. Few of the numbers are making Asterisk 1.6 (using Elastix 2) to report "all circuits are busy now". All of this numbers are cellular phones, but they constantly reporting the same thing. But most numbers even with such music, works well. What do you think I should

Hello, I have MeetMe rooms generated dynamically and it always have two people inside that are entered by dialplan. I wish to make in some way a timeout mechanism that after X amount of time, it will disconnect the users and kick them out of the conference. How can I do such thing ? Thanks, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: