Codec G729 to G711 Outbound

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.

Sounds like a problem with your SIP Trunking provider. Have you contacted them?

Also, you’re more likely to get assistance if you provide us with the following:

  1. Which version of Asterisk are you using?
  2. Which version of FreePBX are you using?
  3. Did you use a Distro? If yes, which one?
  4. You might also have some success on the Elastix forums.