hello i have problem
hi
i am connected to provider thru SIP trunk, with OpenSIPS, on OpenSIPS is connected Asterisk as Endpoint
for SIP phones
everythin is ok while i use ulaw or alaw,
on asterisk: disallow=all, allow=alaw,allow=ulaw
on some phones i need lower codec, so i set up for this extension
disallow=all
allow=g729
no i suppose asterisk should do transcoding, but he don’t, is there any special configuration
so to trunk i’ve add allow=g729 (to PEER and USER detail)
with this configuration, outgoing calls are ok
but inbound call aren’t working, if i understad debug messages correctly, asterisk is trying to make channel to endphone on alaw not with g729 (ist only allowd codec in setup for this sip extensions)
can i change order of preffered codecs in any way ?