Le dimanche 15 novembre 2009 à 23:45 +0100, Eric van der Vlist a écrit : > Weirdly, they seem to be coming from the context I am using to define > outgoing calls rather than the one for ingoing ones (like in asterisk > 1.4), but I guess that's another issue!
Hmmm... I wonder where it can be documented but there seems to be a significant difference in the way I need to write my sip.conf in asterisk 1.4 and 1.6. In 1.4 I had to create two peers (one for outgoing and one for incoming calls). In 1.6 it appears that it just ignores the second peer and that I have to define only one peer for both directions (at least this is working for me). Fully solving my issue involved "consolidating" these two peers into one and adding an "insecure=port,invite" declaration. Eric -- Eric van der Vlist <[email protected]> Dyomedea (http://dyomedea.com) _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
