[Asterisk-Dev] sip from asterisk uses a non-negotiated codec

[Wolfgang S. Rupprecht]

Observed on this morning's CVS-HEAD.  If I enable all the working
codecs in sip.conf's general section as such:

disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=adpcm
allow=g726
allow=ilbc
;; allow=lpc10  (robotman)

And then call do a triangle call from BT100 phone#1 on FWD to asterisk
to BT100 phone#2 I see asterisk using GSM encoding.  This is even
though the phone never mentioned that it understood GSM.

Amazingly the BT100 is able to do *something* with the GSM audio
stream, but the quality is very poor and scratchy.  Shouldn't asterisk
be taking the union of the phone's codec's and it's own?  Strangely
"sip channel show" shows ulaw for both sip channels.  The input from
the phones is ulaw, it is just the output from asterisk that is gsm.

Has anyone else noticed this problem with their asterisk?  

I don't think the FWD leg is significant, but I don't want to wear of
the phone's flash by bouncing it back and forth to often.

-wolfgang
-- 
Wolfgang S. Rupprecht                http://www.wsrcc.com/wolfgang/