[Asterisk-Dev] DTMF signals lost when originating call from a .call file

[Dorian Logan]

Sorry for the repost - I did not give a very good subject line on the 
original posting.

We are creating a .call file to create a call, this call contains a 
voice menu that uses DTMF to select from a set of predefined options. 
The call is placed using an IAX connection to a service provider.

The call works as expected except that the DTMF signals seem to get 
lost. I set the codecs to be alaw and ulaw - but I can not see any 
further opportunities for giving any DTMF settings.

If the menu is called directly via a SIP phone or if it is called from 
an external line (via the same service provider) the DTMF works fine.

Any ideas?

(Just moved to V1.0.2 - problem seems to be the same).

D.
__________________________
e: dorian at tuxstar.com