CallTrex Personal Assistant
2003-Aug-08 12:34 UTC
[Asterisk-Users] queue / agent documentation
We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon
Hi, As far as I know docs are still under the construction and what is available on the sites is it! Senad -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of CallTrex Personal Assistant Sent: 08 August 2003 20:35 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] queue / agent documentation We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
http://www.digium.com/asterisk_handbook/agentlogin_queues.html And you should take a look at queues.conf for some comments detailing the various queue distribution algorithms, ringall, roundrobin, leastrecent so on so forth. -Matt McAughan -----Original Message----- From: CallTrex Personal Assistant [mailto:devon@calltrex.com] Sent: Friday, August 08, 2003 02:35 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] queue / agent documentation We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030808/b6c5e5b4/attachment.htm
My configuration is with a X100P (incoming) and TDM400P w/ 2 ports (agents) and the calls will distribue just as perscribed with ringall and leastrecent. Those are the only two I have used thus far. CVS was a check out from last night. -----Original Message----- From: Brian West [mailto:brian@bkw.org] Sent: Friday, August 08, 2003 03:16 To: 'asterisk-users@lists.digium.com' Subject: RE: [Asterisk-Users] queue / agent documentation> And you should take a look at queues.conf for some comments detailing the > various queue distribution algorithms, ringall, roundrobin, leastrecent so > on so forth.I wanna see if anyone else has seen this result? All except of which do not work. The only method I can get working is ringall. queues.conf: [techsupport] music = default strategy = roundrobin context = demo timeout = 15 retry = 15 maxlen = 0 extensions.conf: [agentlogin] exten => 800,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM}) exten => 800,2,Playback(agent-loginok) exten => 800,3,Hangup exten => 801,1,RemoveQueueMember(techsupport|SIP/${CALLERIDNUM}) exten => 801,2,Playback(agent-loggedoff) exten => 801,3,Hangup [callqueue] ; sales exten => 900,1,Answer exten => 900,2,Queue(techsupport|TtH) exten => 900,3,WaitMusicOnHold(20) exten => 900,4,Voicemail2(u900) exten => 900,5,Playback(vm-goodbye) exten => 900,6,Hangup Here is what I see when doing a roundrobin with two agents added via AddQueueMemeber: It never sends a call to agent 1234 at all... and will only ring 1236 over and over even during a call. I setup iaxclient to load the queue with 4 calls. Then setup two xten sip phones for the agents. During my testing even other routing methods deliver similar results. Bug report is open. http://bugs.digium.com/bug_view_page.php?bug_id=0000045 asterisk*CLI> sip show peers Name/username Host Mask Port Status 1236/1236 65.38.28.149 (D) 255.255.255.255 5060 Unmonitored 1235/1235 (Unspecified) (D) 255.255.255.255 0 Unmonitored 1234/1234 65.38.28.150 (D) 255.255.255.255 5060 Unmonitored asterisk*CLI> show queues techsupport has 4 calls (max unlimited) in 'roundrobin' strategy Members: SIP/1234 has taken no calls yet SIP/1236 has taken no calls yet Callers: 1. IAX[brian@65.38.28.149:5036]/24 (wait: 0:18) 2. IAX[brian@65.38.28.149:5036]/25 (wait: 0:17) 3. IAX[brian@65.38.28.149:5036]/26 (wait: 0:16) 4. IAX[brian@65.38.28.149:5036]/27 (wait: 0:14) default has 0 calls (max unlimited) in 'ringall' strategy No Members No Callers -- Called 1236 -- SIP/1236-7fc3 is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-df37 is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-c56b is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-e9a7 is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-b4dd is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-029d is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-f0aa is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-c264 is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-bd12 is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-197e is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-2056 is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-fa07 is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-994b is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-220d is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-997a is ringing -- SIP/1236-997a answered IAX[brian@65.38.28.149:5036]/24 -- Stopped music on hold on IAX[brian@65.38.28.149:5036]/24 -- Called 1236 -- SIP/1236-e4ab is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-9ed1 is ringing -- Nobody picked up in 15000 ms -- Called 1236 -- SIP/1236-b26c is ringing -- Nobody picked up in 15000 ms == Spawn extension (demo, 900, 2) exited non-zero on 'IAX[brian@65.38.28.149:5036]/24' -- Hungup 'IAX[brian@65.38.28.149:5036]/24' -- Called 1236 -- SIP/1236-bfd6 is ringing _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030808/fde86cb1/attachment.htm