disa

[vortex1]

Anyone that could help on the above?

 

[udosw]

Anyone that could help on the above?

Vortex, please stick to the rules of this forum: Pushing your own articles without any content is not allowed!

To enable DISA in Asterisk on FBF is extremly simple:
First get the module you need (app_disa.so) and put it in usr/lib/asterisk/modules.
Instructions, as well in English, are here:
http://www.ip-phone-forum.de/showpost.php?p=582190&postcount=1
(read befor you ask!)

Create a context like this:

[callthrough]
exten => s,1,NoOp(DISA)
exten => s,n,Answer  
exten => s,n,Playback(beep)
exten => s,n,DISA(no-password,from-intern)

from-intern should be the context your SIP phones normaly dial from.

Udo

 

[vortex1]

Hi.
i did manage to authenticate mayself through DISA, but then the call ends.
here is my config:
sip.conf
[fxogw]
type=peer
fromuser=fxogw
fromdomain=192.168.x.x
auth=md5
secret=xxxxxx
host=192.168.x.y
port=5061
qualify=yes
nat=no
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=info
context=POTS-incoming
insecure=very
callerid=



extensions.conf

[POTS-incoming] ;disa
exten => s,1,Answer()
exten => s,n,Authenticate(123)
exten => s,n,DISA(no-password,from-internal)
exten => s,n,Busy(3)


i am getting :

# asterisk -rvvv
Asterisk 1.4.22, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.22 currently running on fritz (pid = 1028)
No entry for terminal type "vt102";
using dumb terminal settings.
Verbosity was 2 and is now 3

-- Executing [s@POTS-incoming:1] Answer("SIP/192.168.x.y-005c4cc0", "") in new stack

-- Executing [s@POTS-incoming:2] Authenticate("SIP/192.168.x.y-005c4cc0", "123") in new stack

-- <SIP/192.168.x.y-005c4cc0> Playing 'agent-pass' (language 'en')

-- <SIP/192.168.x.y-005c4cc0> Playing 'auth-thankyou' (language 'en')

-- Executing [s@POTS-incoming:3] DISA("SIP/192.168.x.y-005c4cc0", "no-password|from-internal") in new stack
-- Message check requested for mailbox /folder INBOX but voicemail not loaded.
== Spawn extension (POTS-incoming, s, 3) exited non-zero on 'SIP/192.168.x.y-005c4cc0'

fritz*CLI>

 

[vortex1]

update

i have changed the dtmfmode=rfc2833 and i have added relaxdtmf=yes.
even though i am receiving the call to the asterisk and i get a welcome note and it asks the passwd. then i add the passwd, it says thank you, then i get a dial tone (only one) and then it hangs the line.
any ideas on that?