[asterisk-users] basic sip quesiton

[Thomas Perron]

What am I missing please?   sip show registry shows that I am registered.

[general]
register => 5552530146:tamara at sip3.voipvoip.com
;
;
[sip3.voipvoip.com]
bindport=5060 ;you can use different port if the default is blocked
bindaddr=0.0.0.0 ;binds to all



;this is for codec negotiation between the useragent and asterisk
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm



context=incoming ;default context where incoming calls are passed. this
should be the context where your local user.s extensions reside

[outbound-trunk]
;this is the second section of you sip.conf file. Here you can create your
trunk through which you will throw your outgoing calls to axvoice.
host=sip3.voipvoip.com
type=peer
dtmfmode=rfc2833
canreinvite=yes
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