[asterisk-users] Gizmo Dial Out No CALLED PARTY AUDIO??

[Rob]

Yes ... as a matter of fact here is the sip.conf ... obviously private info
removed ....
[general]
register => 1747xxxxxxx:xxxxxxxxxxxx at proxy01.sipphone.com<1747xxxxxxx%3Axxxxxxxxxxxx at proxy01.sipphone.com>
port = 5060
bindaddr = 192.168.22.5
context = incoming
svrlookup=yes
;dtmfmode=inband
allow=all
externip=76.98.xxx.xxx
localnet=192.168.22.0/255.255.255.0

[proxy01.sipphone.com]
nat=yes
;type=peer
type=friend
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
dtmfmode=rfc2833
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
;insecure=very deprecated; use insecure=port,invite instead
insecure=port,invite
qualify=yes
secret=XXXXXXXXXXXX
authuser=1747XXXXXXX
fromuser=1747XXXXXXX
username=1747XXXXXXX
canreinvite=no


On Wed, Aug 5, 2009 at 6:07 AM, Administrator TOOTAI <admin at tootai.net>wrote:

> Rob a écrit :
> > Hi all,
> >
> Hi
> > I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a
> > while and it works fine .... I just added CALL OUT ... I have no problem
> > with call setup ... the called party hears me ... but I can't hear them
> ....
> > again if the call comes INTO the server both sides work fine.
> >
> >
> Looks like a nat issue: do you have nat=yes and canreinvite=no in your
> sip.conf for Gizmo5?
>
> --
> Daniel
>
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