In call DTMF through SIP Proxy/Tunnel connection

[amos.max]

Hi All,
I'm trying to get DTMF to work though the Sip Proxy Manager/Tunnel connection.

Setup:
v10 SP5 (build 22686), Grandstream 4104 PSTN Gateway
Internal Phones: Aastra 480i
External Phones: GrandStream GXP2110, 3CX Softphone

DTMF works from the internal phones fine, but not from the GrandStream or from the softphone.
What's strange is that dialing voicemail and entering the pin works through the SPM/tunnel connection, but in-call DTMF to "outside" parties doesn't.

Since the internal phones work it doesn't seem to me to be a configuration issue with the PSTN gateway, but I have to confess I don't know where to look.

The only thing I can see in the logs is this:

MediaServer|MediaServerReporting:TMFhandler:[MS211000] C:16.1: http://www.xxx.yyy.zzz:10016 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.

Both the GrandStream and the Softphone are set to RFC2833 DTMF with "Payload Number" set to 101, which matches the DTMF payload type in the PSTN gateway.

Any suggestions or pointers are much appreciated!
Rgds - M.

 

[amos.max]

Update:
adding/changing the DTMF method on the GXP2110 and the Softphone to SIP INFO seems to correct the problem.

Is this/what is the recommended configuration for tunnel connections? Only SIP INFO or both SIP INFO and RFC2833?

Thx - M.

 

[leejor]

When generating DTMF outbound through a gateway, the most reliable method is SIP messaging (matching the method used in both the set and gateway) which allows the gateway to re-generate the tones to send out on the phone line. This is especially important if you are using a Codec other than G711 as most others have a problem passing audio DTMF accurately. Going over the Tunnel should not introduce any additional issues.