[Asterisk-Users] DTMFMODE with grandstream

[Rich Adamson]

No, I don't know why. But, the dtmf mode used between the phone and asterisk
stays the same regardless of where you call. That part of the call doesn't
change. Once asterisk has the dtmf digit(s), the next channel (eg, zap, iax)
will use whatever "it" deems to be the correct dtmf mode (unless you have
canreinvite=yes and are using sip on that next leg). That second leg does
not have to be the same dtmf mode as the first leg.

------------------------

> 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.
> > 
> > 
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> 
> 
> 
> 
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