Re: [asterisk-users] Queue Agent Getting Additional Calls When on the Phone
Sorry for the top-post... If you do a core show application AddQueueMember from the cli, you'll see the option I was referring to. You'll also need to make sure you're properly reporting device state to asterisk. I think this means you need to set a call-limit for each sip peer that you want to monitor in sip.conf (we use 25 so there are no accidental limits actually applied), and setup hints in your extensions.conf for each peer. Thanks, --Warren Selby On Oct 14, 2010, at 11:36 PM, Matt Darnell mattdarn...@gmail.com wrote: Warren, I tried using AddQueueMember to add agents. If they a user is on a call asterisk shows: Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers We are using 1.4.36. What did you use to keep track of the extension state? Didn't see any option for that at http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember Thanks for the help. -Matt On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnell mattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Queue Agent Getting Additional Calls When on the Phone
You'll also need to make sure you're properly reporting device state to asterisk. I think this means you need to set a call-limit for each sip peer that you want to monitor in sip.conf (we use 25 so there are no accidental limits actually applied), and setup hints in your extensions.conf for each peer. Warren, Setting the call limits was my issue. I am on a test machine and didn't have it set. Thanks for the help! -Matt -- _ -- 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 Agent Getting Additional Calls When on the Phone
15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? Thanks, --Warren Selby On Oct 14, 2010, at 11:36 PM, Matt Darnellmattdarn...@gmail.com wrote: Warren, I tried using AddQueueMember to add agents. If they a user is on a call asterisk shows: Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers We are using 1.4.36. What did you use to keep track of the extension state? Didn't see any option for that at http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember Thanks for the help. -Matt On Thu, Oct 14, 2010 at 6:04 PM, Warren Selbywcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnellmattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Queue Agent Getting Additional Calls When on the Phone
On 10-10-15 04:10 AM, Сикорский Сергей wrote: 15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter=yes in sip.conf. Leif. -- _ -- 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 Agent Getting Additional Calls When on the Phone
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-10-15 04:10 AM, Сикорский Сергей wrote: 15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter=yes in sip.conf. Leif, Isn't callcounter for 1.6 and not for 1.4? -Matt -- _ -- 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 Agent Getting Additional Calls When on the Phone
2010/10/15 Matt Darnell mattdarn...@gmail.com: On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-10-15 04:10 AM, Сикорский Сергей wrote: 15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter=yes in sip.conf. Leif, Isn't callcounter for 1.6 and not for 1.4? If you are using the Local channel, look into the n option. -M -- _ -- 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 Agent Getting Additional Calls When on the Phone
We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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 Agent Getting Additional Calls When on the Phone
What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnell mattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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
Re: [asterisk-users] Queue Agent Getting Additional Calls When on the Phone
Warren, I tried using AddQueueMember to add agents. If they a user is on a call asterisk shows: Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers We are using 1.4.36. What did you use to keep track of the extension state? Didn't see any option for that at http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember Thanks for the help. -Matt On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnell mattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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 -- _ -- 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 Agent Getting Additional Calls When on the Phone
On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Here is little more console output: localhost*CLI queue show Sales Saleshas 0 calls (max 10) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers localhost*CLI core show channels Channel Location State Application(Data) SIP/101-000b s...@macro-tl-userexten Up VoiceMailMain(101) 1 active channel 1 active call 'core show channels' show SIP/101 is use but 'queue show' does not. -Matt -- _ -- 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
