Jim Singh wrote:
I'm from an SNMP background, so thats the way I'd be looking.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
Something like:
Enable SNMP on your DSL router
Select your scripting language with SNMP support (scotty, shell with net-snmp, python,perl
whatever)
Write a script that queries the router, checking
1) outbound queue lenght,
2) outbound packets/sec
3) interface status
4) dropped udp packets
also perhaps the ping your VOIP provider
check to see if all of the above are within acceptable limits (tweaking required)
Get the script to set a value in the asterisk db based on the result of the check.
get cron to call the script every 60 secs (or whatever)
In your dialplan check the value of the variable and dial outbound calls accordingly.
Tim.
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I just calculated the maximum bandwidth that I wanted to allow for VOIP, divided that buy the amount required per call (ulaw 64k plus overhead - see wiki for examples), fingured I could support "x" number of calls.
Then I used the group function in the dial plan.
-- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
