!I already had the problem last year, then it would be solved (surely from some technician by Deutsche Telekom on their servers), and now I have the problem again (but I didnt changed my Asterisk configuration).The problem: after 15 minutes will ..
!I installed Hylafax on a Ubuntu-Server 14.04. On this server runs Asterisk 11.7.0, too and it was configured like my own Asterisk server at home, but it does not work… :(So, I configured Asterisk to connect to Deutsche Telekom and it does!Then I configu..
!On an Asterisk-Server I have some users. Just two of them have a Mailbox. I want to write a little Web interface to manage many things and Id like to have a menu point for the voicemail, but just if the user has a Mailbox.I found the AMI-Command MailboxStat..
!I have a Server with Ubuntu 14.04 and Asterisk 11.7.0. Id like to have all users on the Active Directory of the network, so that Ican manage all in a central repository.Is it possible? How can I do that?Thanks Luca Bertoncello(lucabert@lu..
and happy new year!My question:- two extensions: 1111 and 2222- an active call on 1111- incoming calls to 1111 should be forwarded to 1111 (call advice!) and 2222I know how can I forward an incoming call to more than an extension,but I have no idea ..
again!With the call pickup-function I can now pickup a call directed to another phone in my Asterisk. Very nice. My problem, now, is that I cant see on my phone, that the other phone (in another room) rings.Is it possible to signal the incoming c..
!Right now I configured my Asterisk to forward the calls for the number X to both phones (mine and the phone of my wife). It works, of course, but Im not enthusiast…I see what we have at office: if one phone rings, other phones in the same group ..
!My Problem: all calls to international numbers will be dropped after exactly15 minutes… I have a VoIP-account by Deutsche Telekom. This is what I see when I call someone (my parents) and the connection will be dropped:== Using SIP RTP CoS mark ..
!My problem: I have three extensions in my Asterisk 126.96.36.199 and they have a voicemail. On two of these numbers the voicemail works without any problem, on the other it doesnt… I get this error:[Oct 17 17:01:29] WARNING: channel.c:5254 set_form..
!I have an Asterisk 188.8.131.52-1 installed on an OpenWRT-Switch. It works correct since July with two VoIP-phones.Now I configured my mobile phone (Samsung Galaxy 2 with Android 4.1.2) to use my Asterisk, so that I can call and receive calls from my mob..