Re: [asterisk-users] Queue - how to jump to next member after NO ANSWER?
On 23 July 2013 23:18, Shishir Pokharel shishir.pokha...@on24.com wrote: Read queue configuration esp. QEUUESTRATEGY and agent TIMEOUT. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jarek Jarzebowski *Sent:* Tuesday, July 23, 2013 3:04 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Queue - how to jump to next member after NO ANSWER? ** ** Hi all, I have a Queue with 3 members: SIP/100 SIP/200 SIP/300 When call arrives SIP/100 is ringing.. After given timeout ringing stops but call is not routed to next member but SIP/100 starts ringing again.*** * I know that this is because SIP/100 is still available in the Queue but is it any way to make a Queue witch strategy: call SIP/100 - if it is BUSY, UNAVAILABLE, PAUSED or _NO_ANSWERED_ after given number of time - hump to the next member? Thanks in advance. Jarek -- _ This is in the queues.conf in 1.8 If you want the queue to avoid sending calls to members whose devices are known to be 'in use' (via the channel driver supporting that device state) uncomment this option. (Note: only the SIP channel driver currently is able to report 'in use'.) ringinuse = no -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue - how to jump to next member after NO ANSWER?
Hi all, I have a Queue with 3 members: SIP/100 SIP/200 SIP/300 When call arrives SIP/100 is ringing.. After given timeout ringing stops but call is not routed to next member but SIP/100 starts ringing again. I know that this is because SIP/100 is still available in the Queue but is it any way to make a Queue witch strategy: call SIP/100 - if it is BUSY, UNAVAILABLE, PAUSED or _NO_ANSWERED_ after given number of time - hump to the next member? Thanks in advance. Jarek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue - how to jump to next member after NO ANSWER?
Read queue configuration esp. QEUUESTRATEGY and agent TIMEOUT. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jarek Jarzebowski Sent: Tuesday, July 23, 2013 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queue - how to jump to next member after NO ANSWER? Hi all, I have a Queue with 3 members: SIP/100 SIP/200 SIP/300 When call arrives SIP/100 is ringing.. After given timeout ringing stops but call is not routed to next member but SIP/100 starts ringing again. I know that this is because SIP/100 is still available in the Queue but is it any way to make a Queue witch strategy: call SIP/100 - if it is BUSY, UNAVAILABLE, PAUSED or _NO_ANSWERED_ after given number of time - hump to the next member? Thanks in advance. Jarek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
