Hey. I was trying for several weeks to get my asterisk to work with voip with audio in both directions. I feel there must be a better way to have my configuration.
As you can see externaddr is not set and is the only way I get 2 way audio. Otherwise i get 1 one or none.
[code][general]
context=incoming ; Default context for incoming calls. Defaults to 'defau$
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
allowguest=no
;disallow=all ; First disallow all codecs
;allow=g729 ; Allow codecs in order of preference
;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packe$
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
session-timers=refuse
register => XXXXXXXXX:XXXX@sip.flowroute.com
localnet=192.168.77.0/255.255.255.0
;externaddr=190.86.XXX.XXX
nat=force_rport,comedia
directmedia=no ; Asterisk by default tries to redirect the
registertimeout=15
registerattempts=0
;progressinband=yes
[flowroute] ;keep this lowercase, do not change format
type=peer
secret=XXXXXXXXXXXX
defaultuser=XXXXXXXX
;username=XXXXXXXX
host=sip.flowroute.com
;dtmfmode=rfc2833
context=default
directmedia=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
insecure=invite
;insecure=port,invite
fromdomain=sip.flowroute.com
trustrpid=yes
sendrpid=yes
[/code]
Any suggestions/reason why externaddr has to not be set in order for audio to work.
If your interested I can post sip traces or anything else you would like.