Re: [Asterisk-Users] FWD, DISA DTMF

I've solved. I've putting rfc2833 also on SIP client that connect to first asterisk. Igor On Mon, 2004-07-12 at 22:53, Igor Barsanti wrote: 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

[Asterisk-Users] FWD, DISA DTMF

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 nothingany tips ??? Thanks, -- Igor

Re: [Asterisk-Users] SIP vs. SIP :-(

Resolved... canreinvite=no (i've put careinvite :-)) igor On Tue, 2004-06-01 at 19:24, Igor Barsanti wrote: 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

[Asterisk-Users] SIP vs. SIP :-(

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

[Asterisk-Users] SIP 404 error....

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 = _X.,1,Dial,SIP/[EMAIL PROTECTED]|60|r When i try to call a number i

[Asterisk-Users] Call monitoring

I start cal monitoring with: exten = _1XX,1,Answer exten = _1XX,2,Monitor,wav exten = _1XX,3,Dial(SIP/${EXTEN}|30|tr) I can record the call, that is correctly forwarded to SIP destination, but i cannot ear the ringing tone. If i put exten = _1XX,3,Dial(SIP/${EXTEN}|30|mr) i can ear instead

Re: [Asterisk-Users] Gatekeeper

]) Igor Barsanti wrote: I've setup an asterisk server H.323 compliant, with a GnuGk gatekeeper. My h323.conf is: [General] port=1720 gatekeeper=127.0.0.1 context=default [PABX] type=H323 e164=PABX prefix=0 context=default my gatekeeper.ini contain: [RasSrv::GWPrefixes] PABX=0

[Asterisk-Users] Gatekeeper

I've setup an asterisk server H.323 compliant, with a GnuGk gatekeeper. My h323.conf is: [General] port=1720 gatekeeper=127.0.0.1 context=default [PABX] type=H323 e164=PABX prefix=0 context=default my gatekeeper.ini contain: [RasSrv::GWPrefixes] PABX=0 When i call, from a gatekeeper

[Asterisk-Users] H.323, video and asterisk....

Just few question about H.323 1) I can authenticate H.323 users without a Gatekeeper 2) If i have two asterisk server, connected with an IAX2 trunk, an H.323 client on the server 1 can make video call to an H.323 client on server 2 ??? 3) An H.323 client can make a video conference

[Asterisk-Users] IAX and Voicemail

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