Been working at getting ADA working with asterisk 1.6 and freepbx 2.5 and have hit a number of hurdles. I'll post the things I had to do to make things work below as a few things are not in the documentation and I have it MOSTLY working.
The one remaining issue I have is that if I am on a call on my phone, and I idly use ADA to make a call, my phone returns busy then sends the call to voicemail & rings the destination. The end result is that the other party either gets the end of a leave your message call or just silence as asterisk waits for a message. Normally this wouldn't be an issue but I have had several phones hang sip channels so that they can make calls but the pbx sees the phone as busy.
How do I stop asterisk from finishing the call if the phone is busy?
changes made (needs to be done for all extensions):
manager_custom.conf
[XXX]
secret = PASSWORD
deny=0.0.0.0/0.0.0.0
permit=A.B.C.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,all
write = system,call,log,verbose,command,agent,user,all
(please note the requirement of "all" or originate for asterisk 1.6 and XXX is your extension, PASSWORD is your password and A.B.C is your ip range)
create users.conf
[XXX]
fullname = FULLNAMEHERE
hasvoicemail = yes
secret = PASSWORD
vmsecret = 1234
hassip = yes
hasmanager = yes
extensions_custom.conf (this is the bit that needs work)
[ada]
;exten => 200,1,Dial(Local/200@from-internal,,i,)
;exten => 200,2,Hangup()
include => from-internal
reload asterisk and you should be able to connect with the client with extension@ipOfServer and your password with no problems.
as you can see I've tried a couple of things (the other 50 aren't shown
any help would be great as this is pretty much the final hurdle to having a nice elegant solution for click2dial in asterisk ... well this and a tapi driver compiled for 64bit (come on guys the siptapi team have done it
[Edit fix glaring spelling mistakes]
All the best.
Kactus
