Hello,
I want to send calls from Asterisk server to another SIP server. I want to through calls directly without user/pass authentication and registration. How should configure sip.conf for this ? Please someone help me.
Thanks in advanced.
I don’t fully understand the question, but:
allowguest=yes (which is default) in the general section will cause incoming calls to be passed to the context specified in the general section (default context is “default”), as long as they don’t match a more specific entry.
A peer entry matching the IP address of the source, and with no secret, or with insecure=invite, will accept all calls with their last hop at that address.
I think the same applies to user entries, with no secret, if the From header (or is it Contact?) matches the sip.conf section name. I presume insecure=invite works there, as well, although it is normally only used for peer mode entries.
On the outgoing side, you will need to specify the host name or address, if there is no registration.
(In case you want Asterisk to act as a proxy, it won’t. All calls must be terminated and then chained with Dial, etc.)
This was the wrong forum. I have answered assuming that you meant to use Asterisk Support (i.e. you are not trying to do this through one of the Asterisk GUIs).
Hi,
What i was looking for is incoming SIP call authenticated by IP address only. Like the way VoIP wholesalers do. I got the solution here,
freepbx.org/forum/freepbx/us … -using-fre
sip.conf should be like this,
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
host=x.x.x.x
insecure=very
port=5060
qualify=yes
type=friendinsecure=very no longer does anything, and insecure=invite is not useful if you don’t have a password.
type=friend means anyone who knows the sip.conf section name can all through this entry without having to match the host address. It is rarely needed. Almost always type=peer is better.
As a double check, you should explictly restrict the IP addresses allowed.