On Asterisk 1.4, Call doesn't line Channel: A&B. You could put the second dialplan snippet into a context and do your callfile like this: [callccm] exten => s,1,Dial(SIP/CCMMAIN,10,KkTt) Exten => s,n,Dial(SIP/CCMSLAVE,10,KkTt)
-- Channel: SIP/104 CallerID: SIP/104 MaxRetries: 1 WaitTime: 60 retryTime: 5 Context: callccm Extension: s -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, October 15, 2009 7:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP and call manager > > Here are two ways to address this > > 1. Dial(SIP/CCMMAIN&SIP/CCMSLAVE) - this tries both at once > > 2. exten => s,1,Dial(SIP/CCMMAIN,10,KkTt) > Exten => s,n,Dial(SIP/CCMSLAVE,10,KkTt) > > CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3 > rings) > > Danny thats good to know for extensions.conf but I am using call files. echo "Channel: SIP/CCMMAIN/5551212" > /tmp/call echo "Context: smvoice-test" >> /tmp/call Can I do the Channel: SIP/CCMMAIN/5551212&SIP/CCMSLAVE/5551212 in the Channel for the call file? Jerry _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users