I’m try to arrange a small time self-help system i.e. people call in, press the desired number and get instructions. Since this is for a non-profit org, the free part is quite important.
Here’s my setup:
I have a Google Voice number which I have forwarded to a Gizmo account (G1) which forwards all calls another Gizmo account (G2, through SIP). Asterisk registers G2.
Problem: The initial message plays fine when the user calls through a cell phone. The initial message asks the user to press 1,2,3 etc. Asterisk does not receive the tones for the pressed extension.
i.e. Google Voice => G1 => G2 (Asterisk) = extensions dialed not picked up.
I tried this using G1 to G2 call only. The message plays fine, and the selected extension is recorded by asterisk and the correct help file is played.
G1 => G2 (Asterisk) = extensions dialed are picked up.
I am guessing the dtmfmode value in incorrect. I tried rfc2833, inband, sip-info, auto values. Same result. I also tried switching the value of canreinvite to yes.
sip.conf entries:
[general]
dtmfmode=rfc2833
context=help-test
srvlookup=yes
nat=yes
externip=24.xxx.xxx.xxx
localnet=192.168.1.0/255.255.255.0
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
qualify=no
defaultexpirey=3600
register=1747xxxxxxx:my_secret@proxy01.sipphone.com
canreinvite=yes
[proxy01.sipphone.com]
type=friend
context=help-test
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
dtmfmode=rfc2833
host=proxy01.sipphone.com
insecure=port,invite
secret=my_secret
username=1747xxxxxxx
fromuser=1747xxxxxxx
fromdomain=proxy01.sipphone.com
canreinvite=no
qualify=no
extensions.conf
[help-test]
exten=s,1,Wait(1)
exten=s,2,Answer()
exten=s,3,Wait(1)
exten=s,n(lbl_help-test_0),Background(help-start)
exten=s,n,WaitExten(15)
exten=s,n,Hangup()
exten=i,1,Playback(pbx-invalid)
exten=i,n,Goto(s,lbl_help-test_0)
exten=t,1,Playback(pbx-goodbye)
exten=t,n,Hangup()
exten=1,1,Playback("hello-world")
Problem 0:
I have to make that double Gizmo jump because Google Voice => Asterisk (As gizmo number) simply didn’t work. It crashes midway through the dialplan.
Asterisk version : 1.4.18.1 running under VMWare