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
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