search for: barsanti

I can dial from an asterisk host to another one via FreeWorldDialup, on the other side DISA service answer to me and i can ear dialtone. But i cannot send DTMF and dial an extension on the DISA enabled asterisk.....i've tried rfc2833 and inband...but nothing....any tips ??? Thanks, -- Igor Barsanti GPG Public key available at http://pgp.mit.edu http://pgp.mit.edu:11371/pks/lookup?op=get&search=0xD29D4C21

I'v a sip client and a sip trunk to FWD: [general] port=5060 context=default tos=reliability disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid="Me" host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of

I've setup a simple asterisk test environment with an ISDN card configure in modem.conf and a gnophone client connected to my asterisk server via IAX. I can place call and answer, i've also succesfully configured a voicemail. The problem is that i cannot redirect call to my voicemail when gnophone isn't active. The call will be redirected when the timeout end, i want this only if

I've buyed SIP traffic from platinumcalling.com.au. in sip.conf i have: [platinum] context=default type=peer username=MYUSERNAME secret=MYPASSWORD host=sip.platinumcalling.com.au in my extension.conf: [default] exten => _XXXXX.,1,Dial,SIP/${EXTEN}@platinum|60|r When i try to call a number i got: Got SIP response 404 "Not Found" back from 203.30.19.164 SIP/platinum-67f2 is