[Asterisk-Users] Out Of Band DTMF and SIP

[Clif Jones]

I am currently using Asterisk with G.711 codecs and in-band DTMF for
several Cisco 7960's
and an Audiocodes GW.  When allowing out-of-band DTMF, I could use
voicemail menus and
anything else on Asterisk that required DTMF but I could not get the
DTMF relayed out of the
GW.  Has anyone verified that this works between 2 SIP devices?  If so,
I would be interested
in your settings. Also, I would really like to know what debug level to
use (if any) that would allow
me to see that the Phone Event codec packets are being relayed from the
Cisco to the GW.
Finally, if the GW was unable to convert phone events to DTMF tones,
will Asterisk generate
the tones on the GW call leg if I configure the SIP phone for
out-of-band DTMF and the SIP
GW for in-band?  Thanks for your help!



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