Yusuf wrote: > Hi Dan, > I used asterisk 1.2.10 with asterisk-addons 1.2.3. > I did two successfull calls, but with dtmf=rfc2833, dtmf was not > sending at all. Then when I made some changes, I could not get any > calls to go through. The call would just hangup after first ring. Call Manager's support for RFC2833 is 'lacking'. It works reasonably well in 5.X for SIP, but forget about using it with H323. I've tested all four options with Call Manager, and only q931keypad and h245signal worked. I'd recommend using h245signal.
> Did you get calls going in both ways, inbound and outbound to asterisk. > O got two calls going from CAllmanger to asterisk only, other would not > work. Calls work both ways, although 99.99% of my calls are inbound, since we use Asterisk for conferencing only at this point. Here are a couple of ideas to try: 1. Set the Call Manager H323 gateway to 'Require MTP' 2. Set DTMF to h245 signal What is likely happening is that with Asterisk asking for RFC2833, CCM tries to invoke a MTP. I am not sure in which Asterisk-Addons version it was added, but I wrote support for Empty Terminal Capability sets for chan_ooh323. If that feature is not in the version you have, (chan_ooh323 release 0.5 or newer), and you are not forcing an MTP on the CCM gateway you'll see a problem like you have. I should also point out that if you are not running chan_ooh323 0.5 or newer and do get Asterisk to accept calls with out forcing an MTP, calls will be dropped anytime a CCM endpoint uses hold or transfer features. If you do have 0.5 or newer, then changing the DTMF migth be enough. Hope this helps, Dan _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users