On Mon, Jan 23, 2017 at 6:38 PM, Fred Muteesa <[email protected]> wrote: > Hello Dev team, > > I have been playing with asterisk dial function and I have the senarial > below. > > I am generating a call from server 1 and receiving it on server 3, but I > want server 1 to control how long this call should be. > > Though I placed server 2 in the middle which is able to modify my parameters > of the dial function and control call duration. > > How do I detect on server 1 that server 3 has hangup so that server 2 does > not keep the call connected longer than I require. > > This is of extreme importance to me all advise and help will be appreciated. > > > > > > On Server 1 > > [to_server2] > > exten => 1234,1,Dial(SIP/server2/1234,3,S(3)) > > exten =>1234,2,Hangup() > > > > on Server 2 > > [from_server1] > > exten => 1234,1,Dial(SIP/server3/1234,,gS(15)) > > exten =>1234,2,wait(15) > > > > on Server 3 > > [from_server2] > > exten =>1234,1,answer() > > exten =>1234,2,wait(3)
Hey Fred, First off, welcome to the Asterisk Development mailing list, Fred! This list is typically used for C code level discussion of Asterisk and its associated modules and internals. Sometimes telecommunications protocol level discussions occur as well. For dialplan and related questions such as this, I would recommend using the asterisk-users mailing list or the discourse forums on community.asterisk.org. I would think that your dialplan setup should allow server 1 to detect when server 3 hangs up as long as you indeed have server 1 dialing server 2, and server 2 dialing server 3. When you post your question to the -users list, I would also include the output of the asterisk verbose logs from the server that the hangup does not happen on as expected. You might also enable sip debug, using 'sip set debug on' on it so that a protocol level dump is available. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
