[asterisk-users] Problem setting for incoming termination

[Olle E. Johansson]

12 aug 2011 kl. 14:51 skrev Kevin P. Fleming:

> On 08/11/2011 02:03 AM, Jim Boykin wrote:
> 
>> We have difficulty setting up the incoming termination for our
>> clients. Both the ends are using asterisk.  The problem is unless we
>> use fromuser at client end, it does not work properly as expected.
>> 
>> Below is a configuration at our end. The problem is that whenever call
>> is received from the client, it goes to default context instead of
>> 'dallas' context. Also, the ${CDR(accountcode)} variable remains
>> empty. Now, If we set fromuser field at the client end, then
>> everything starts working, however, in that case, it overrides the
>> callerid.
> 
> This is a known and well-understood problem caused by the method that Asterisk users for SIP authentication; the 'From' header in the incoming INVITE is used *both* for determining which user is placing the call and for Caller ID. As you note, if you have the real Caller ID in that header, then Asterisk can't use it for matching to a user in sip.conf, and thus can't authenticate the call properly.
> 
> The solution for this is to use 'sendrpid' on the sending end and 'trustrpid' on the receiving end; this will configure Asterisk to transfer the Caller ID information in a Remote-Party-ID (or P-Asserted-Identity, depending on the version you are using) header, allowing the From header to be used solely for authentication.

Or stop using type=user and type=friend, and stick to type=peer and ASterisk will only match on IP+port address.

/O