hi i have tested it with sip info option in grand stream as DTMP relay and dtmfmode=rfc2833 and it works , that's ok but problem is this i cant ask users change their DTMP on their ip phones, so i should use auto on asterisk to detect who is comming with which DTMF mode, i change dtmfmode in asterisk to auto and i have Context: giti Nat: RFC3581 DTMF: auto Qualify: 0 Use ClientCode: No
asterisk says , dtmfmode=auto : Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones if the remote side does not indicate support of rfc2833 in SDd i have tested it , dtmfmoed=auto in sip.conf and dtmf mode inband in grandstreamm and evene teletronics and it dosnt work . do you know why ? thanks giti Rich Adamson <[EMAIL PROTECTED]> said: > > > i have GXP-2000 ( grandstream ) and and i am trying to press key fron phone > > keypad when i hear greating message and asterisk asks me select one > > extention ( i have backgroud function in my extentions.conf ) , > > with grandstream asterisk dosnt receive anything from ip-phone , but with > > same test with wiztel wifi phone , i have incoming key in asterisk and > > extention selects and ..... > > > > i have this sip config on my asterisk : > > Global Settings: > > ---------------- > > SIP Port: 5060 > > Bindaddress: 192.168.0.19 > > Videosupport: No > > AutoCreatePeer: No > > Allow unknown access: Yes > > Promsic. redir: No > > URI user is phone no: No > > Our auth realm asterisk > > Realm. auth: No > > User Agent: Asterisk PBX > > MWI checking interval: 10 secs > > Reg. context: (not set) > > Caller ID: asterisk > > From: Domain: > > Record SIP history: Off > > Call Events: Off > > IP ToS: 0x0 > > OSP Support: No > > SIP realtime: Disabled > > > > Global Signalling Settings: > > --------------------------- > > Codecs: ulaw,alaw,ilbc > > Relax DTMF: No > > Compact SIP headers: No > > RTP Timeout: 0 (Disabled) > > RTP Hold Timeout: 0 (Disabled) > > MWI NOTIFY mime type: application/simple-message-summary > > DNS SRV lookup: Yes > > Pedantic SIP support: No > > Reg. max duration: 3600 secs > > Reg. default duration: 120 secs > > Outbound reg. timeout: 20 secs > > Outbound reg. attempts: 10 > > > > Default Settings: > > ----------------- > > Context: giti > > Nat: RFC3581 > > DTMF: info > > Qualify: 0 > > Use ClientCode: No > > Progress inband: Never > > Language: (Defaults to English) > > Musicclass: default > > Voice Mail Extension: asterisk > > > > > > > > and this is my extentions.conf : > > > > > > exten => 1019,1,Wait,1 ; Wait a second, just for fun > > exten => 1019,2,Answer ; Answer the line > > exten => 1019,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds > > exten => 1019,4,ResponseTimeout,10 ; Response Timeout to 10 seconds > > exten => 1019,5,BackGround(/etc/asterisk/giti) ; a congratulatory message > > > > I don't have a GXP-2000 to test with, but most sip phones will not send > any dtmf unless you press the # key after the digit, or wait for the > phone's built-in timer. So try 4# (or whatever digit) to see if that has > an impact. > > If the GXP-2000 has an option to set dtmf to rfc2833, use that instead > of "info", and include "dtmfmode=rfc2833" in the extension definition > in sip.conf. > > > _______________________________________________ > --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 > _______________________________________________ --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
