On 31 Dec, 04:02, phpguy <ksma...@hotmail.com> wrote:[color=blue]
> i got an asterisk@home 2.7 box setup with zap and voipbuster
>
> the zap works and i set it up to dial using my landline and fxo card
> when i dial 99= something it ommits the 99 and dials the rest
>
> i want to set up the voipbuster trunk to do the same with 88 but for
> some reason it wotn connect
>
> i used the default settings im behiond a router and i put nat to yes
> but nothing
>
> what do i need to do ? all my phones are sip phones linksys spa 921
>
> i can see from the flash panel its going through to trunk but nothing
> happens then times out, i cant hear anytihng
>
> i setted up asterisk as sip trunk not iax2
>
> thanks in advance
>
> OUTGOING SETTINGS
>
> host=194.120.0.198
> nat=1
> secret=XXXXXX
> type=peer
> username=XXXXXXXX
>
> //////////////////////////////////
>
> INCOMING SETTINGS
>
> context=from-pstn
> secret=XXXXXXXXXXXXXXX
> type=user
>
> /////////////////////////////
>
> REGISTRATION
>
> XXXXX:XXXXXXX...@194.120.0.198
>
> ////////////////////////////////
>
> SIP.CONF
>
> ; Note: If your SIP devices are behind a NAT and your Asterisk
> ; server isn't, try adding "nat=1" to each peer definition to
> ; solve translation problems.
>
> [general]
> port = 5060 ; Port to bind to (SIP is 5060)
> bindaddr = 192.168.100.69 ; Address to bind to (all addresses on
> machine)
> disallow=all
> allow=ulaw
> allow=alaw
> context = from-sip-external ; Send unknown SIP callers to this context
> callerid = Unknown
>
> #include sip_nat.conf
> #include sip_custom.conf
> #include sip_additional.conf
>
> //////////////////////////////////////[/color]

It is terrible. I would like to suggest you try miniSipServer. It is
very easy to use.