[Freeswitch-users] Setting codec/dtmf mode

[David Aldworth]

I'm looking for the most effective way to make sure I'm always forcing  
inband dtmf and PCMU on the PSTN <-> FS side of inbound and outbound  
calls. FS is always in the middle of the media. The FS <-> SIP UA  
(customer) side will be rfc2833 and whatever the negotiated codec for  
that particular UA happens to be. I know I can set <param name="codec- 
prefs" value="PCMU"/> and <param name="inbound-codec-negotiation"  
value="greedy"/> in the internal sip profile but won't the external  
sip profile settings override this when UA dial out? (they hit the  
external profile first in this case)

I'm basically fishing for suggestions on the best way to use start/ 
stop_dtmf for the inband detection and start/stop_dtmf_generate for  
sending the dtmf.

In asterisk this would have been accomplished by setting up separate  
stanza's in sip.conf and setting the dtmfmode= and allow= line per the  
respective legs of the calls. So, calls coming to/from the PSTN would  
have dtmfmode=inband and allow=ulaw, meanwhile UA's connecting to  
asterisk would have dtmfmode=rfc2833 and allow=ulaw, gsm, etc.

Why on earth would I be doing this? Well, in the interest of keeping  
the explanation short, we are limited to the common denominator of all  
our upstream PSTN carriers and they (or their equipment rather) always  
support this setup.

Thanks for any advice.

David