Queue will continue if called person hangs up (and there's no option). If caller hangs up, call goes to h extension in same context. Just the same way as Dial with 'g'. There's a change in 1.6 that allows called channel to continue if caller hangs up, so probably something like this could be applied also to Queue (or was that actually working with using Local channels?).
Regards, Atis On Wed, Apr 23, 2008 at 7:13 PM, AnDY <[EMAIL PROTECTED]> wrote: > Thank you for your answer. > But the Dial command has a option 'g' which means that after succes will > proceed next priorities in the dialplan. Is there something also for > Queue() because according to manual there is no option for it. So I am > looking for some other solution. > > Andy > > Tony Mountifield napsal(a): > > > > In article <[EMAIL PROTECTED]>, > > <[EMAIL PROTECTED]> wrote: > > > >> Hello everybody. > >> > >> I was looking for the solution but nothing found. I have this in my > >> extensions.conf: > >> > >> exten => 233,1,SetAccount(queue1) > >> exten => 233,2,Queue(queue1|rn) > >> exten => 233,3,NoOp(${QUEUESTATUS}) > >> exten => 233,4,NoOp(${DIALSTATUS}) > >> > >> > >> But when the call is placed in the queue and somebody answer it, it will > >> throw an error: > >> == Spawn extension (default, 211, 4) exited non-zero on > >> 'Local/[EMAIL PROTECTED],2' > >> > >> And no other command in extensions is executed. > >> Any suggestions? > >> > > > > Queue() is like Dial(), in that if it succeeds in connecting to someone, > > it will not return to the next priority in the dialplan. However, if you > > define an 'h' extension, that will get executed when the call is complete: > > > > exten => 233,1,SetAccount(queue1) > > exten => 233,2,Queue(queue1|rn) > > exten => 233,3,NoOp(${QUEUESTATUS}) > > exten => 233,4,NoOp(${DIALSTATUS}) > > > > exten => h,1,NoOp(${QUEUESTATUS}) > > exten => h,2,NoOp(${DIALSTATUS}) > > > > Cheers > > Tony > > > > > > > _______________________________________________ > -- 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 > -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 _______________________________________________ -- 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