[Asterisk-Users] DTMFMODE with grandstream

Rich Adamson radamson at routers.com
Mon Dec 19 06:14:08 MST 2005


>   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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