[Asterisk-Users] DTMFMODE with grandstream

[giti at dataproducts.ae]

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 <radamson at routers.com> 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
>