[Asterisk-Users] inband dtmfmode, SIP to VoicePulse, > 1 digit extentions do not work?

[James W. Brinkerhoff]

I have a situation where calls come in via SIP from VoicePulse and get dropped 
into a main menu.    Voicepulse only works /w dtmfmode=inband and I only 
allow ulaw/alaw as codecs.

When the call comes in and gets dropped to the menu, you can hit 1, 2 or 3 to 
get to other people.   Or you can dial the persons full extention... 1000 
1001 etc...     /w dtmfmode=inband, it only ever grabs the first digit rather 
than collecting "tones" till it hits the DigitTimeout.

I tried from a SIP hardphone that uses rfc2833 and it worked as expected 
(collected digits until the timeout, then tried routing).

Anyone know WHY this happens and if there is some sort of workaround?   
Barring that, does anyone know a SIP or IAX provider that will give me 
inbound/outbound calling /w a 212 areacode phone number ( ie + 1 212 - 
XXX-XXXX ) and uses dtmfmode = info or rfc2833 ?

Thanks a bunch everyone,

-jwb