I'm trying to bundle the powers of Asterisk and SER. Asterisk for pabx functionalities and termination to landline/PSTN, and SER as SIP Gateway/Proxy. With my current configuration the SIP user just adds 0 as a prefix to a number, and the call will go out to PSTN over Asterisk. For this to work I added the rewritehostport() function in SER to point to the Asterisk IP (different from the SER ip). At the moment I just added the following line to my sip.conf (in the [general] section): context=from-sip But my question here is, everyone can (ab)use this by connecting directly to the Asterisk IP. This way they can easily dial out over the PSTN network.
Hi,
This sounds a very similar problem to me, despite the different context.
The 'default' context in the [general] section shouldn't be (ab)usable - set this to something like [bogon-calls].
Then set up a specific peer lower down:
[ser] context=sip-legal host=y.y.y.y ; IP address of SER
Se this Wiki page for more flesh of my (not yet fully working!) configs: http://voip-info.org/wiki-Asterisk+cisco+FXO
Good luck! Fran. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
