Re: [asterisk-users] question on SIP and call manager
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
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
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) -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 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on SIP and call manager Customer has 2 call manager systems and I am using asterisk to place calls through the CCM. One for the main use - CCMMAIN and another for disaster CCMSLAVE. Can asterisk be setup in such a way that calls first try to use CCMMAIN and if thats not available use CCMSLAVE. Example if I place a call file that places a call like Dial: SIP/CCMMAIN/5551212 that if CCMMAIN is not available then CCMSLAVE will automatically be used? My application placing calls in the call file doesnt have any knowledge of which context to use. CCMMAIN is the only thing my call file nows about. How do I set up such an arrangement if possible? thanks. 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
[asterisk-users] question on SIP and call manager
Customer has 2 call manager systems and I am using asterisk to place calls through the CCM. One for the main use - CCMMAIN and another for disaster CCMSLAVE. Can asterisk be setup in such a way that calls first try to use CCMMAIN and if thats not available use CCMSLAVE. Example if I place a call file that places a call like Dial: SIP/CCMMAIN/5551212 that if CCMMAIN is not available then CCMSLAVE will automatically be used? My application placing calls in the call file doesnt have any knowledge of which context to use. CCMMAIN is the only thing my call file nows about. How do I set up such an arrangement if possible? thanks. 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
