It looks like there is some bugs with the DISA function. I tried a fresh new SME 7.0 R2 installation with these packages:
smeserver-asterisk-1.2.3-2.i686.rpm
smeserver-asterisk-zappri-MPP-1.2.2-1.i686.rpm
smeserver-asterisk-sounds-1.2.2-2.noarch.rpm
Configuration/use of the DISA function is just stright forward:
(extensions.conf)
exten => 1,1,Authenticate(123123123)
exten => 1,2,DISA(no-password|trunkname)
This should normally be working, but it does not. It should give back a stable dial tone to the dialer (??) but it just give some unstable "wawa's" and then the caller is auto logged off after a few seconds. Messages from the Asterisk CLI:
sme7*CLI>
sme7*CLI>
-- Playing 'auth-thankyou' (language 'no')
-- Executing DISA("SIP/12345678-bd0d", "no-password|trunkname") in new stack
== Spawn extension (incomming, 1, 2) exited non-zero on 'SIP/12345678-bd0d'
sme7*CLI>
Unless I do something wrong (I don't know about), I believe that this is an error that will repeat on all installations.
By the way - It works 100 % OK for the "ordinary telephone functions" and there is no problems (that I have fount until now) dealing with ordinary incomming and outgoing telephony functions.
(I wonder if I red that the newest A@H has some problems with the DISA function as well. Could it have something to do with the newest Asterisk server itself ?)
And one other thing - I think that it is a right approach to do a Astereisk impementation to the SME server to chose a rather "minimalistic approach" like these Aterisk rpms. To bundle something like A@H will destroy your SME installation completely. I think that if "the Asterisk part" of the SME server should have a graphical interphace at all, it should be something like a easy test editor for the configuration files, like the Astlinux
http://www.astlinux.org/(Reason: When configuring Asterisk there is so many factors outside the server itself that you will have to take into consideration. If you make an easy standard configuration via a graphical shell a la server-manager this might fit with one vendor and one kind of equipment, but not all of them. Because of this there is no way around to let the user manually edit the configuration files itself as long as the sip/iax technology is just so variated and unstandardized like it actually still is today. Editing the asterisk configuration files could be done via ssh shell or via a simular web shell a la Astlinux. With such a "minimalistic aproach" it should be possible to run the Asterisk process(es) in the sme box without messing it all up. )
Best reg Arne.