> In our setup, outbound call volume frequently exceeds the > line capacity of the DSL line. We do not want to move to > another codec to better utilize the line, but instead wish to > automatically divert overflow to the Long Distance T1 when > the DSL is "full". Ideally the system would also be able to > adjust automatically to network conditions such as network > outage, high latency, jitter and/or packet loss. If the LD T1 > was also full or if there was no other path, Dial would > return busy/congestion instead of connecting a call of low quality. > > I realize that one solution is to manage variables using > macros in the dialplan and keep a count of VOIP calls. I > believe that this a) difficult to maintain b) can be > difficult to dynamically adjust based on parameters from the > jitter buffer, round trip time, and/or packet loss c) > couldn't be "the best way to do it". > > Before I go slinging code, does anyone know of a clean > solution? Do other people need / desire this functionality? > > Our Setup: > Software: Suse 9.2 + Asterisk 1.0.7 (built from CVS) > Network: DSL measured to be 2 mbps up / 430 kbps down > Termination: IAX2 / G711 / nufone and voipjet > Zaptel: 2 digium 100 cards one connected to a Siemens > PBX and the other to a (long distance) provider,signaling is E&M Wink >
Jim: I am doing something very similar using SetGroup and CheckGroup. Take a look at those commands on the wiki, and especially to the n + 101 priority to accomplish a failover in the case that your "trunk limit" is reached. Marty _______________________________________________ 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
