I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488
web admin page you enter these registration values. When you reboot the
HT488 you should see it registering on Asterisk CLI.
What's left is a dialplan line in extensions.conf like this:
exten => 9,1,Dial(SIP/<sip acount name>,10)
I've tried your example shown here. When I dial 9 I get dial tone from
the HT488 but I when I try to dial, nothing happens (i.e. I keep hearing
dial tone even though I'm dialing). Any ideas?
That may be related with the dtmfmode. Can you try inband? I believe rfc2833
should work too, but once you have it working with inband, you can test the
rest.
Also I think you'd like to use PCMU codec on HT488, other codecs may cause
DTMF detection problems (iLBC seems fine though).
In short, I would play with DTMF and codec parameters on both sides.
Hope this helps,
Soner
_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
