Friends,
I’m new to Asterisk and still being apredizagem, set up a lab and I’m having a following problem and would like your help to get through.
I have the following scenario.
(| SIPFONE | (G.729) <- sip —> (G729 | ASTERISK | G711 )<— trunksip —> (ulaw | cisco | ulaw )—>( IP_PHONE).
1.When run a call internally between branches using different codecs (g.729/ulaw) can hear and talk on both sides.
2.When I run an external connection, through my sip trunk using the ulaw codec, I can hear and talk on both sides.
3.When I run an external connection, through my sip trunk using G.729 codec, I can hear perfectly, plus, on the other side has much noise, low volume, and many cuts in his voice.
I have this set in my trunk.
PEER DETAILS:
type = friend
qualify = yes
nat = no
insecure = very
host = x.x.x.x (IP do Callmanager)
fromdomain = x.x.x.x (IP do Callmanager)
dtmf = auto
disallow = all
context = from-trunk
canreinvite = yes
allow = ulaw
Thank you.