Well it's not working as it should. Every call go to Dail(SIP/sip1) and if no one respond then to the next one :( Arkon
----- Original Message ----- From: "Atis Lezdins" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Friday, November 23, 2007 3:06 PM Subject: Re: [asterisk-users] Check if SIP is avaible to dial > Jakub Syrek wrote: >> I thing there was an error in last version of my macro, correct one (i >> hope): > > Just test it :) > >> >> [macro-call] >> ;sip1 - firs channel from sip outgoing cals operator >> ;sip2 - second channel from sip outgoing cals operator >> ;sipn - N channel from sip outgoing cals operator >> ;ARG1 - outgoing telephone number >> exten => s,1,SetGroup(${ARG1}) >> exten => s,2,GotoIf($[ ${GROUP_COUNT(${ARG1})} < 2 ]?dial-ok) > > I don't really get why you have this. Why set group count on called > number? You won't be able to call the same number two calls at time, you > really want that? There are lot of numbers, that can accept unlimited > amount of calls simultaneously. > >> >> ;outgoing number was called previously by someone else? >> exten => s,n,NoOp(-- Call destination was previously called and its >> busy --) >> exten => s,n,Hangup >> >> ;we can dial >> ;now chack witch chanel is free to meake a call with >> exten => s,n(dial-ok),SetGroup(sip1) >> exten => s,n,GotoIf($[ ${GROUP_COUNT(sip1)} > 1 ]?sip1-busy) >> exten => s,n,Dial(SIP/sip1/${ARG1},25,m) >> exten => s,n,Hangup > > You should set group after checking that trunk is available. Once group > on channel is set, it's not automatically removed, it remains set until > channel is alive (or you manually unset - i just don't remember the > syntax). > >> >> exten => s,n(sip1-busy),SetGroup(sip2) >> exten => s,n,GotoIf($[ ${GROUP_COUNT(sip2)} > 1 ]?sip2-busy) >> exten => s,n,Dial(SIP/sip2/${ARG1},25,m) >> exten => s,n,Hangup >> >> ;and so on to last one >> >> exten => s,n(sipN-busy),SetGroup(sipn) >> exten => s,n,GotoIf($[ ${GROUP_COUNT(sipn)} > 1 ]?all-busy) >> exten => s,n,Dial(SIP/sip2/${ARG1},25,m) >> exten => s,n,Hangup >> >> ;every channel is busy >> exten => s,n(all-busy),Hangup > > Regards, > Atis > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
