Yesterday I rebooted the server and it seems to be working again. Not sure what the reboot might have changed. Hopefully it doesn't happen again but I can't be sure. To answer your question I have the sip.conf in my mysql database and in MySQL I have callcounter set to yes. I don't have a column of 'qualify' in my database for the sip users. For my config I am using OpenSIPS as the register and proxy. Asterisk is only used for voicemail and ACD/Hunt groups.
On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot <[email protected]>wrote: > > Provide the entry for Agent SIP/9013XX9XX8 along with parameters > 'callcounter' and 'qualify' from sip.conf. > > Also provide CLI outputs of 'core show channels',sip show peers' and 'queue > show' when... > > (1)First caller enters the Queue > (2)First caller gets connected with Agent > (3)First caller gets disconnected from Agent > (4)Second caller enters the Queue > > You may have sequences changed for step no 3 and 4 in your scenario. > > > [SATISH] > > On Sat, Jun 11, 2011 at 2:56 AM, <[email protected]> wrote: > >> Queue not sending call to Agent >> >> >> >> I am having an issue and i am not sure if it is a bug or a config issue. I >> was originally running Asterisk 1.8.1.1 when I noticed this issue. I >> upgraded to 1.8.4.2 to see if that would fix it but it didn't. >> >> The issue is that I have a call queue and the agent dials a number to log >> into the queue. When someone calls the queue the first time the call is sent >> to the agent without issue. The issue is that any calls after the first are >> placed in the queue and never sent to the agent who is logged in and >> available. Before I call the queue I do a "show queue" and it shows the >> agent as >> >> Asterisk18*CLI> queue show >> irock.com has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, >> 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s >> Members: >> SIP/9013XX9XX8 (dynamic) (Not in use) has taken no calls yet >> No Callers >> >> >> Then the call comes into the queue and the callee just sits in the queue. >> When I do a "show queue" again when the callee is in the queue it shows the >> agent as busy >> Asterisk18*CLI> queue show >> irock.com has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime, >> 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s >> Members: >> SIP/9013XX9XX8 (dynamic) (Busy) has taken no calls yet >> Callers: >> 1. SIP/9013XX9XX8-00000001 (wait: 0:12, prio: 0) >> >> >> So I am not sure what happened because the agent was free before the call. >> If I do a reload at the Asterisk CLI and then call again the agent gets the >> call and then the second call is once again placed in the queue. I will >> attach a SIP Debug that shows what is going on. I don't see any SIP invites >> leaving Asterisk to invite the agent to the call. >> >> One other thing.... Currently in my config I have the agent show up as >> just the username which is the phone number. If I set it so that the agent >> shows up as phonenumber@blah then I can call the agent constantly without >> any issue. The only problem here is that when I do a "queue show" the agent >> shows up as "unknown" status. So when the agent is on a call and someone >> else calls the agent will be interrupted. >> >> >> >> This is what I have in queues.conf >> [irock.com] >> strategy=ringall >> ringinuse=no >> joinempty=yes >> leavewhenempty=no >> announce-frequency=30 >> min-announce-frequency=15 >> periodic-announce-frequency=60 >> announce-holdtime=yes >> announce-position=yes >> >> ; ("You are now first in line.") >> queue-youarenext = queue-youarenext >> ; ("There are") >> queue-thereare = queue-thereare >> ; ("calls waiting.") >> queue-callswaiting = queue-callswaiting >> ; ("The current est. holdtime is") >> queue-holdtime = queue-holdtime >> ; ("minutes.") >> queue-minutes = queue-minutes >> ; ("seconds.") >> queue-seconds = queue-seconds >> ; ("Thank you for your patience.") >> queue-thankyou = queue-thankyou >> ; ("Hold time") >> queue-reporthold = queue-reporthold >> ; ("All reps busy / wait for next") >> periodic-announce = queue-periodic-announce >> >> >> >> This is what I have in extensions.conf >> exten => 9012XX1XX1,1,Answer() >> exten => 9012XX1XX1,n,Set(QUEUE_MAX_PENALTY=0); >> exten => 9012XX1XX1,n,Queue(irock.com,t) >> exten => 9012XX1XX1,n,Hangup() >> >> exten => *50,1,Answer >> exten => *50,n,Macro(queue-login,${EXTEN},${EXTEN:0:4}) >> exten => *50,n,Hangup >> >> exten => *51,1,Answer >> exten => *51,n,Macro(queue-logout,${EXTEN},${EXTEN:0:4}) >> exten => *51,n,Hangup >> >> [macro-queue-login] >> exten => s,1,Set(agent=${EXTEN:4}) >> exten => s,n,Set(queue=irock.com) >> exten => s,n,NoOp(Queue login agent ${EXTEN:4} to queue ${phone}); >> exten => s,n,AddQueueMember(${queue}); >> exten => s,n,Playback(agent-loginok) >> >> [macro-queue-logout] >> exten => s,1,Set(agent=${EXTEN:4}) >> exten => s,n,Set(queue=irock.com) >> exten => s,n,NoOp(Queue logout agent ${EXTEN:4} from queue ${phone}); >> exten => s,n,RemoveQueueMember(${queue}); >> exten => s,n,Playback(agent-loggedoff) >> -- >> _____________________________________________________________________ >> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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 > -- -- *--*--*--*--*--* Duane *--*--*--*--*--* --
-- _____________________________________________________________________ -- 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
