[asterisk-users] Interruptible announcements in queue application
Hello all, Ive found another issue with the queue application. Assuming Ive configured a queue with a long periodic announcement and have two queue members assigned. Both queue members are busy at a time, while another caller is joining the queue. After a while the periodic announcement is played back to the caller, in that case it takes about 40 seconds to be played back. If then one of the two agents becomes available, the call is unfortunately not routed to the agent, until the playback of the announcement has finished. If you display the agent status and the queue to a supervisor he can see that there are callers waiting up to 40 seconds, even if there are available queue members. For inbound call centers tat is more than suboptimal. Does anybody know if somebody already created a patch to interrupt queue announcement when an agent becomes ready? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Backports to 1.2.14 of 1.4.0 app_queue features.
Hello Gavin,
Ive found a small bug in your patch,
/* If the queue entry is within avl [the number of available members] calls
from the top ... */
if (ch idx avl) { -- Bug!!!
if (option_debug)
ast_log(LOG_DEBUG, It's our
turn (%s).\n, qe-chan-name);
/* If the queue entry is within avl [the number of available members] calls
from the top ... */
if (ch idx = avl) { change here
if (option_debug)
ast_log(LOG_DEBUG, It's our
turn (%s).\n, qe-chan-name);
Otherwise if you have 3 queue members ready, and 3 waiting callers, only 2
instead of 3 phones are ringing at the same.
One other question, Ive recognized that after applying the patch, the queue
application is distributing the calls using last in, first out strategy.
So if there is a large queue, the first callers in the queue are never
distributed to a queue member. Do you know anything about that behaviour,
Im not sure if it was the same before applying the patch?
Thanks and Regards
Markus
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[asterisk-users] Re: Delay in Call Distribution using the Queue Application
Thanks for the info, is there a patch available for version 1.2 that adds the autofill option? Thanks and Regards Markus Yes, I confirm the autofill option is present in 1.4, but must be enabled manually not to break compatibility with 1.2. l. On Fri, 19 Jan 2007 15:32:32 +0100, Tom Rymes [EMAIL PROTECTED] wrote: You may be running into the limitation in Asterisk 1.2 (It's fixed in 1.4, I think double check that) in how the queues distribute calls. Basically, the queue can only distribute one call at a time, so if you have two agents, both available, and two calls in the queue, asterisk will send call #1 to agent #1 first. Once that call is connected, Asterisk will then send call #2 to agent #2. In other words, until asterisk distributes the first call, it can't distribute any other calls waiting in line. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay in Call Distribution using the Queue Application
Hello all, we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent. For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the wrapuptime parameter in queues.conf to 0, for my point of view an agent phone that becomes available again should ring immediately after hanging up a call. Does anybody know if there are any known issues or restrictions in the queue application in version 1.2.12.1? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk IAX Trunk and Queues
Hello all, If I have connected 2 asterisk boxes over an IAX trunk, how can I forward a call in the dial plan from one machine to the other? If I'm using call queues, is it possible to add queue members that are not local but configured in the second asterisk machine, which is connected via the AIX trunk? If no, is there any concept of global queuing, which means to have one Asterisk working as dispatcher distributing calls to extensions or agents on other asterisk boxes connected via AIX trunk? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] average waiting time in a queue
Hello all, we want to use asterisk queues for a call center application. Depending on the average waiting time in a queue, we want to make a decision to either enqueue a call or transfer it to another site. Are the applications available to query the average waiting time of a queue, if possible for a configurable time frame? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Log out an Agent on RNA
Hello all, Is it possible to automatically log off an agent on RNA (Ring No Answer) when the agent is logged in with AgentCallbackLogin? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WG: Asterisk and Agents
Hello NG,
We've a small problem using agents in asterisk. One requirement is, if there
no agent logged into a queue, it shouldn't be possible that a call joins a
queue. I can configure that using the parameter joinempty=strict in
queues.conf, unfortunately the parameter takes only effect if I add
members to the queue dynamically. If there are static members assigned to
the queue, a call can always join the queue, even if there are no agents
logged. To add agents dynamically to a queue I'm using the following scripts
in the dialplan:
exten = _*8XXX,1,Answer
exten = _*8XXX,2,SetLanguage(de)
exten = _*8XXX,3,AddQueueMember(DEMO|Agent/${EXTEN:1})
exten = _*8XXX,4,Dial(Local/999/n,,D(#))
exten = _*8XXX,5,AgentCallBackLogin(${EXTEN:1}|[EMAIL PROTECTED])
exten = _*8XXX,6,Hangup()
exten = _**8XXX,1,Answer
exten = _**8XXX,2,SetLanguage(de)
exten = _**8XXX,3,RemoveQueueMember(DEMO|Agent/${EXTEN:2})
exten = _**8XXX,4,AgentCallbackLogin(${EXTEN:2})
exten = _**8XXX,5,Hangup()
So if I type e.g. *8000 it logs in Agent/8000 and adds the agent dynamically
to the queue test. With **8000 it logs out Agent/8000 and removes the agent
from queue test. All that work's fine. The problem is that to add an agent
to a queue, it has NOT to be defined in agents.conf. If an agent mistypes
his agent ID, e.g *8999, it logs on Agent/8999, even if it is not defined in
agents.conf. As result Agent/8999 keeps assigned to the queue DEMO, and
because there is an agent assigned the parameter joinempty has no effect
anymore. Calls can join the queue even if there is no real Agent logged in.
Any ideas are welcome.
Best Regards
- Markus
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[Asterisk-Users] Timeframe for QueueStatus values
Hello all, I've a question regarding the values completed and abandoned that are returned by the manager command queuestatus. What is the timeframe for these values, are they counted since the last asterisk boot, or per day, or is the timeframe configurable? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Agents
Hallo all, I've a question regarding the agent concept of asterisk. If I login an agent (using AgentLogin), this agent is directly ready to receive calls. From the most other ACD-systems I know that an agent first logs into the system and then has to set himself ready to receive calls. So the most common agent states are login, ready, not ready, wrapup and logoff. How is ready/notready/wrapup implemented in asterisk? Thanks and Regards Markus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Test environment for a Predictive Dialer
Hello all, I'm thinking about to set up a test environment for a predictive dialler with two asterisk machines. Each Asterisk should use a Digium TE110P card. One machine should work as predictive dialler; the other box should simulate the PSTN. - Is it in general possible to interconnect the two asterisk machines in that way? Do I need any hardware in between to connect the two TE110P cards? - Can I simulate the PSTN with a Digium TE110P card? Thanks and Regards Markus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs
Hello all, I've been using chan_capi-cm-0.6 as CAPI channel driver, the driver was working fine until I've reinstalled asterisk last week. I retrieved the latest asterisk version from cvs and then build and installed it. When rebuilding the capi channel driver with the latest asterisk headers I receive the following warning: ... chan_capi.c:4014: Warning implicit declaration of function use_ast_mutex_init_intstead_of_pthread__mutex_init ... If I start asterisk, it fails with the following error: [chan_capi.so]Nov 5 17:16:51 WARNING[2830]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: use_ast_mutex_init_instead_of_pthread_mutex_init Nov 5 17:16:51 WARNING[2830]: loader.c:554 load_modules: Loading module chan_capi.so failed! Any help would be appreciated! Markus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs
Thanks Armin, this version is working, but I still have an undefined symbol in another module: ... [pbx_wilcalu.so]Nov 5 18:51:12 WARNING[11348]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Nov 5 18:51:12 WARNING[11348]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! ... Can you also help me on that issue? Thanks and Regards Markus -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Gesendet: Samstag, 5. November 2005 18:25 An: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Betreff: Re: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs Please try chan_capi-cm CVS HEAD on sourceforge.net Armin On Sat, 5 Nov 2005 [EMAIL PROTECTED] wrote: Hello all, I've been using chan_capi-cm-0.6 as CAPI channel driver, the driver was working fine until I've reinstalled asterisk last week. I retrieved the latest asterisk version from cvs and then build and installed it. When rebuilding the capi channel driver with the latest asterisk headers I receive the following warning: ... chan_capi.c:4014: Warning implicit declaration of function use_ast_mutex_init_intstead_of_pthread__mutex_init ... If I start asterisk, it fails with the following error: [chan_capi.so]Nov 5 17:16:51 WARNING[2830]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: use_ast_mutex_init_instead_of_pthread_mutex_init Nov 5 17:16:51 WARNING[2830]: loader.c:554 load_modules: Loading module chan_capi.so failed! Any help would be appreciated! Markus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
