Re: [Asterisk-Users] Agents getting logged off agressively
On 1/16/06, Bud Bach wrote: I have a group of agents logged in to a queue that is set for ringall. The agents are set to auto logoff if they don't answer in 15 seconds incase they step away without logging out. That works fine, however, if they are on a call and a new call comes in, they are getting logged out too. The phones are ATA's connected via SIP. One thought is that the phones may be allowing a second call so they really don't look busy and that's why they get logged out. Anyway, is there a way to prevent busy agents from getting logged out? Yes. Use PauseQueueMember and UnpauseQueueMember before and after the call from queue to the agents which will make certain no additional calls get sent to the agents that are already on a call.How do I do this? I currently: exten = s-OPEN,1,Queue(q)to queue a call to an agent. How do I get the agent and queue to pass to PauseQueueMember?Do I just need to pause them for calls (in/out) that were not queued?-- Bud ___ --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] Agents getting logged off agressively
I have a group of agents logged in to a queue that is set for ringall. The agents are set to auto logoff if they dont answer in 15 seconds incase they step away without logging out. That works fine, however, if they are on a call and a new call comes in, they are getting logged out too. The phones are ATAs connected via SIP. One thought is that the phones may be allowing a second call so they really dont look busy and thats why they get logged out. Anyway, is there a way to prevent busy agents from getting logged out? -- Bud ___ --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] Manually Opening and Closing a Queue
Does anyone have a snippet of extensions.conf to share where they call a number to open or close a queue? Thanks. -- Bud ___ --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
RE: [Asterisk-Users] Automatic logoff of all agents at set time
But, if the agents don't log out for some reason, they will still be logged in the next time the queue opens even if they aren't there right? -- Bud -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Tuesday, December 27, 2005 5:27 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Automatic logoff of all agents at set time On 16:22, Tue 27 Dec 05, Chuck Bunn wrote: Hi, Is there a way to force the logoff of all agents at a set time say 8:00PM or do I have to do an agentcallbacklogin with each agents credentials? I am using Asterisk 1.2 The wiki shows an extension that the agent calls to preform the logoff - I need something that is completely automated as we need calls to stop going to a queue and to go to voice mail after hours. Hi, You dont have to logoff your agents to do this. Have a look at the extensions.conf cmd GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime Good luck -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --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 ___ --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] Music On Hold
Help! No Music on Hold. Probably a novice mistake but I cant figure it out. Here are the details: CentOS 4.2 Asterisk 1.2.1 (Do I need to do something to get MOH to build?) Ztdummy loaded (conference works fine) musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 Sip device (x-lite also tried with an ATA) with canreinvite=no: sip.conf: [7211] username=7211 secret= host=dynamic type=friend context=standardphone disallow=all allow=gsm allow=ulaw allow=alaw allow=g723.1 allow=g729 canreinvite=no Extensions.conf: exten = 8702,1,Answer() exten = 8702,n,MusicOnHold(default) exten = 8702,n,Hangup() # asterisk -r Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.2.1 currently running on ccsip (pid = 4782) Verbosity is at least 3 == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-be01' -- Executing Answer(SIP/7211-cedb, ) in new stack -- Executing MusicOnHold(SIP/7211-cedb, default) in new stack -- Started music on hold, class 'default', on channel 'SIP/7211-cedb' -- Stopped music on hold on SIP/7211-cedb == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-cedb' The Stopped music on hold happens immediately like it cant find something. Should I give up and use madplay? -- Bud ___ --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
RE: [Asterisk-Users] Music On Hold
Now Im really baffeled. I found some comments at the end of the musiconhold.config file about the native format. I copied the files in /var/lib/asterisk/mohmp3 to /var/lib/asterisk/moh-native (just cpt them). Then I uncommented the section in musiconhold.config: [native] mode=files directory=/var/lib/asterisk/moh-native and change the dialplan to: exten = 8702,n,MusicOnHold(native) And it works. Now, how do I make native the default? I tried to copy: mode=files directory=/var/lib/asterisk/moh-native from the native section to the default section and that didnt work -- Bud Dec 16 12:17:38 WARNING[6222]: interface.c:215 decodeMP3: Junk at the beginning of frame -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bud Bach Sent: Friday, December 16, 2005 10:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music On Hold Help! No Music on Hold. Probably a novice mistake but I cant figure it out. Here are the details: CentOS 4.2 Asterisk 1.2.1 (Do I need to do something to get MOH to build?) Ztdummy loaded (conference works fine) musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 Sip device (x-lite also tried with an ATA) with canreinvite=no: sip.conf: [7211] username=7211 secret= host=dynamic type=friend context=standardphone disallow=all allow=gsm allow=ulaw allow=alaw allow=g723.1 allow=g729 canreinvite=no Extensions.conf: exten = 8702,1,Answer() exten = 8702,n,MusicOnHold(default) exten = 8702,n,Hangup() # asterisk -r Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.2.1 currently running on ccsip (pid = 4782) Verbosity is at least 3 == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-be01' -- Executing Answer(SIP/7211-cedb, ) in new stack -- Executing MusicOnHold(SIP/7211-cedb, default) in new stack -- Started music on hold, class 'default', on channel 'SIP/7211-cedb' -- Stopped music on hold on SIP/7211-cedb == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-cedb' The Stopped music on hold happens immediately like it cant find something. Should I give up and use madplay? -- Bud ___ --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
RE: [Asterisk-Users] Music On Hold
Thanks Klaus. I missed the make mpg123 step! -- Bud -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Peras Sent: Friday, December 16, 2005 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Music On Hold Wich player do you use? I use the one that is coming with Asterisk. Just cd to the Asterisk Sources, make mpg123, cd mpg..., make make install and i´m done. It worked fine all the time. cheers klaus Bud Bach schrieb: Help! No Music on Hold. Probably a novice mistake but I cant figure it out. Here are the details: CentOS 4.2 Asterisk 1.2.1 (Do I need to do something to get MOH to build?) Ztdummy loaded (conference works fine) musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 Sip device (x-lite also tried with an ATA) with canreinvite=no: sip.conf: [7211] username=7211 secret= host=dynamic type=friend context=standardphone disallow=all allow=gsm allow=ulaw allow=alaw allow=g723.1 allow=g729 canreinvite=no Extensions.conf: exten = 8702,1,Answer() exten = 8702,n,MusicOnHold(default) exten = 8702,n,Hangup() # asterisk -r Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.2.1 currently running on ccsip (pid = 4782) Verbosity is at least 3 == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-be01' -- Executing Answer(SIP/7211-cedb, ) in new stack -- Executing MusicOnHold(SIP/7211-cedb, default) in new stack -- Started music on hold, class 'default', on channel 'SIP/7211-cedb' -- Stopped music on hold on SIP/7211-cedb == Spawn extension (standardphone, 8702, 2) exited non-zero on 'SIP/7211-cedb' The Stopped music on hold happens immediately like it cant find something. Should I give up and use madplay? -- Bud ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
