[Asterisk-Users] DTMF with Asterisk as SIP client

[Yair Hakak]

Hello,
 I have the following setup:

sip phones <->SER <-> asterisk <-> voip provider1
                                                <-> voip provider2
i got a toll-free DID from voipprovider1 to allow people from outside
to call into asterisk, get authenticated, and use voipprovider2 to
call out (kind of a primitive calling card app).

anyway, voiprovider is giving my bad DTMF (usually 2 keypresses for
every 1)...transport is via SIP, i am registered in sip.conf with a
register statement (i.e. asterisk is a SIP client) and ulaw and alaw
are the first allowed codecs. When i set dtmf as info or RFC2833 i
don't get any tones, and when i set inband i'm back to bad DTMF.

if i call into the extension from one of my sip phones (i.e. not via
voip provider) and interact with the menu (put in my authentication
and dial the onward number) it works fine.

anyone come across this? any tips on how to solve it?

any help is appreciated,

 thanks,
 yair