[Asterisk-Users] context being ignored by inbound sip call

[btb]

Johnathan Corgan wrote:
> btb wrote:
> 
>> [7508] ;ipkall
>> type = peer
>> dtmfmode = rfc2833
>> context = remote
>> callerid = "ipkall incoming" <7508>
>> nat = no
> 
> You've configured this entry as a peer, which is for dialing out, versus
> as a user, which is for incoming calls.  Solution is to change to
> 'type=user'.
> 
> If you really need a peer definition, you can use 'type=friend', which
> will cause * to create both a user and a peer entry for '7508' using the
> parameters listed.  Some parameters are common to both peers and users
> so it saves space.
> 
> Personally, I never use the 'type=friend' method, but rather maintain
> separate peer and user sections for outbound and inbound calls to/from
> other switches or endpoints.  This helps _me_ keep things straight;
> others (probably most) prefer the combined 'type=friend' method, though.

thanks jonathan-

i originally had this entry as type=user, and switched to type=peer 
after finding the context was being ignored and reading that type=user 
may/is be(ing) phased out:

http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer

i've tried type=user again (as well as type=peer), with some additional 
parameters (mostly guesses, because i don't yet fully understand 
registration):

[7508] ;ipkall
type = peer
host = dynamic
dtmfmode = rfc2833
context = remote
callerid = "ipkall incoming" <7508>
nat = no
insecure = very

i gather the ideal method is to know the source ip and source port of 
the connection from my peer, and include that in the sip config?  how 
can i make asterisk tell me where a connection is coming from?

-ben