[Freeswitch-users] Problem with outbound calls

[Eduardo Casarero]

Hi list, i'm a brand new freeswitch user (without previous asterisk/voip
experience), after reading all wiki pages, google searchs, etc i need some
help to solve a problem.

configuration:

Freeswitch -> Firewall (nat) -> internet -> Sip Provider

In my current configuration the gateway is REGED and inbound calls (from
provider to freeswitch) works ok with good audio quality. However outbound
calls don't. When i call through the gateway the destination phone rings,
and when is answered there is no audio.

I've check with "show channels" in fs_cli and i cant see any codec in the
read_codec write_codec part, they are blank. I've reviewed all sip profiles
configuration, but obviously i'm missing something.

I will really appreciate any comment,guidance,help,etc. (if someone is in
Buenos Aires/Argentina i can also offer a free beer!)

Thanks in advance.

Eduardo.
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