Hi David, In the scenario you describe, I would expect to see one of the following solutions (but not both at the same time): 1. OpenSIPS acts as the registrar for the SIP phones. Calls (INVITE requests) from SIP phones are routed on via a SIP trunk 2. OpenSIPS acts as a transparent proxy in front of another SIP server such as Asterisk
Scenario 1 is the most common. OpenSIPS authenticates calls based on a list of credentials that it holds, normally in the subscriber table. In this case, you really want to avoid the situation where each outbound call triggers an additional authentication request from the SIP trunk. Can you re-configure your Asterisk endpoint so it trusts INVITE requests coming from your OpenSIPS server? E.g. add the line insecure=INVITE to the sip peer definition. In scenario 2, which I would not consider to be the preferred solution, OpenSIPS just passes the SIP messages between the phone and the Asterisk server - in both directions. OpenSIPS does not authenticate calls because that job is done by the Asterisk server and all the credentials are held by Asterisk, not by OpenSIPS. In this case the 401 request would just be passed upstream to the phone. Try to avoid the situation where OpenSIPS is authenticating the INVITE from the SIP phones using its own list of credentials, but then it also has to authenticate each call sent over the SIP trunk. In theory you could use the UAC_AUTH module of OpenSIPS to do this, but in practice I have never been able to make this work because it breaks the CSeq numbering sequence of the SIP request messages. John Quick Smartvox Limited _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
