guys…
I’ve got several sip phones that support g729 codec annexA/B that register to *now server.
how can I enable this phones to talk with codec g729 in pass-thru mode??
any advice appreciated… 
hi asifsajjad…thanks for reply…
can you explain more detail…
sorry…i’m very new in asterisk…
I don’t have the g729 option at codec preference in user configuration page?
and i only have this is my in sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[authentication]
- should i put this in sip.conf or extension.conf??
another question…
- i also have h323 device with g729 support, i assume
the configuration in h323.conf below is enough to enable pass-thru for h323 device??
type=peer
context=default
ip=192.168.x.x
port=1720
disallow=all
allow=g729
e164=2000
rtptimeout=60
dtmfmode=rfc2833
thanx before…
- should i put this in sip.conf or extension.conf??
another question…
- i also have h323 device with g729 support, i assume
the configuration in h323.conf below is enough to enable pass-thru for h323 device??
type=peer
context=default
ip=192.168.x.x
port=1720
disallow=all
allow=g729
e164=2000
rtptimeout=60
dtmfmode=rfc2833
thanx before…
ok…thanx…asifsajjad 