Dean @ INKnBITs
2006-Jul-18 01:33 UTC
[asterisk-users] SIP ATA Channels for outbound calls - How to select in dialplan
I have setup 3 Linksys SPA-3000 devices to pass/send our analog voice calls into/out of asterisk. The inbound calls work fine as I have set the spa-3000's to forward all calls to an extension. I have added them to the sip.conf as spa-3k1, spa-3k2, and spa-3k3. Is there a way for when some picks up a phone to dial, it starts at 3k1, if congestion, move onto the sk2, and so on. I'm looking for it to find the first available line to use. Is this possible in the dialplan? Thanks, Dean.
Devraj Mukherjee
2006-Aug-28 21:03 UTC
[asterisk-users] SIP ATA Channels for outbound calls - How to select in dialplan
I am not sure if you have solved this already, but this may be something you are interested in [outbound-local] exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9NXXXXXX,2,Congestion( ) exten => _9NXXXXXX,102,Congestion( ) exten => 911,1,Dial(${OUTBOUNDTRUNK}/911) exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911) On 7/18/06, Dean @ INKnBITs <dean.bath@inknbits.co.uk> wrote:> I have setup 3 Linksys SPA-3000 devices to pass/send our analog voice calls > into/out of asterisk. The inbound calls work fine as I have set the > spa-3000's to forward all calls to an extension. I have added them to the > sip.conf as spa-3k1, spa-3k2, and spa-3k3. Is there a way for when some > picks up a phone to dial, it starts at 3k1, if congestion, move onto the > sk2, and so on. I'm looking for it to find the first available line to use. > Is this possible in the dialplan? > > Thanks, > Dean. > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >