There are three choices for Send DTMF: in-audio, via RTP(RFC2833) and via SIP info. It’s set to RTP(RFC2833).
On my PBX in a Flash all extensions are setup with rfc2833 for dtmf mode. My SIP trunks are also set up as dtmfmode=rfc2833.
In my sip_custom.conf there is no reference to rfc2833 - is that correct?
I would need to see your sip_custom.conf before I could tell you if there should or shouldn’t be a reference to RFC2833.
The thing I find very suspicious is that your RTP debug shows no media coming from your phone and being sent to your ITSP.
There are only three strings in my sip_custom.conf
My external IP
My external host
My local network
What’s your opinon as to the quality of the Grandstream GXP2000? Could this be the issue?
I guess what I am not understanding here is why we aren’t seeing the audio from the grandstream in your RTP debug.
The RTP debug should show two way audio
received from public ip
sent to private ip
received from private ip
sent to public ip
right now we are only seeing 1/2 the story. can you enable both “rtp debug” and “sip set debug” and send me the entire transaction which should be 2 INVITE (in/out) ending at the BYE/OK.
I don’t think i had it set correclty to show all the lines of buffer. try this on my pastebin - same location as before.
note that i’ve deleted some lines as it is too large for pastebin. it’s the latest under ruhroh.
ok, that looks better. it does appear that asterisk is receiving the RFC2833 packet and forwarding it out. Your configuration is all correct and the phone is functioning properly and asterisk appears to be functioning properly.
There is one other thing you should try before pointing the finger at your ITSP and saying they aren’t dealing with the DTMF properly. I would suggest upgrading to the most recent version of 1.4 (1.4.24.1 as of today) to see if the problem goes away.
If that doesn’t fix things then it’s time to call your ITSP to see what they say.
Ok - thanks a lot for all your help and persistence with troubleshooting this problem.