Re: [asterisk-users] 20min waiting time
On 8/15/07, OCOSA ListAcct [EMAIL PROTECTED] wrote: Did not work either...Thank you! Otis Michiel van Baak wrote: On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote: exten=5,2,Dial(SIP/supportSIP/support2,2,tr) Make this line read: exten=5,2,Dial(SIP/supportSIP/support2,,tr) That should do the trick As Eric observed in an earlier email in this thread, the phone handset is probably giving up on you and responding with a No-Answer message to asterisk. The only solutions that I am aware of are 1) A queue (as previously suggested) or 2) A dialplan which loops around before the 75 seconds of ringtime are up, hangup the call and dial it again. The queue is the better option IMHO as it allows hold music and announcements. OTOH, it does mean that the caller is paying for an answered call. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Did not work either...Thank you! Otis Michiel van Baak wrote: On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote: exten=5,2,Dial(SIP/supportSIP/support2,2,tr) Make this line read: exten=5,2,Dial(SIP/supportSIP/support2,,tr) That should do the trick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote: exten=5,2,Dial(SIP/supportSIP/support2,2,tr) Make this line read: exten=5,2,Dial(SIP/supportSIP/support2,,tr) That should do the trick -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Eric ManxPower Wieling wrote: Steve Totaro wrote: Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve Sorry to reply to my own post but for clarification, we had four queues. English sales, English support, Spanish sales, Spanish Support. At peek times, we would have 200-300 agents logged in and 600 or so callers. This was usually when our ads were running during Jerry Springer or Judge Judy. I think his two agent single queue would work just fine. Add Queuemetrics which is free (I believe) for five or less agents and then you can actually get some reporting on how your support role is handled. In your situation it seems that queues work well for you. When you have dedicated agents answering calls full time queues work well. In non-call shops people forget to log out of the queue, are away from their desk often, and otherwise just screw up many of the assumptions that the Asterisk queue system makes. This is in addition to the learning curve. For a low number of calls and/or non-dedicated agents, a little bit of dialplan logic can do everything someone needs with something that is massively more flexible. So you are basically saying that Asterisk Queues are not really a bad thing. So your blanket statement, Queues in Asterisk are horrid little creatures. should really read, *Untrained users of Queues in Asterisk are horrid little creatures *Well with any system, if users are not trained properly or do not use the system correctly, there will be problems. I would not blame it on the software through. There are steps you can take to mitigate some of the harm done by user error. One is the removal of an agent from the queue if they do not answer the call (agentcallback), that is built into Asterisk. Another is a more complicated but works well. It involves an ActiveX control loaded into IE that speaks jabber to your Asterisk box. The user/agent must be actively moving through the CRM or a popup will ask if they are still there, they have ten seconds to click yes, or Jabber sends a message to Asterisk to remove them from the queue. Another thought I had but it never went beyond a thought was sort of like reverse answering machine detection. Some sort of logic that listens to the agents channel for voice (such as hello or thank you for calling) within a relatively short period of time. If no voice was detected, the agent would be logged out and the call routed to another agent. This feature would be great for call centers that use agentlogin rather than agentcallback. Tied with Queuemetrics, you can quickly see who is logging in and out excessively and re-train those users on proper use. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Trevor Peirce wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. I wonder if this is your phone deciding it has been ringing for long enough and rejecting the call. That is probably the case. I know that the Linksys ATAs do not ring indefinitely. After exactly 60 seconds of ringing, it sends back a message of: SIP/2.0 480 Temporarily not available, thus causing the dialplan logic to continue on to voicemail. Your best bet is to setup a roundrobin queue and cycle through every 50 seconds or so. Perhaps a NoOp(DialStatus is ${DIALSTATUS}) would shed light on this possibility? Trevor --- Andres http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 20min waiting time
I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Steve do you have an example that works for you. I am reading the queue literature nowThank you! Otis Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Eric so I should do this exten=5,1,Answer exten=5,2,StartMusicOnHold exten=5,3,Dial(SIP/supportSIP/support2,2,tr) exten=5,4,VoiceMail([EMAIL PROTECTED]) exten=5,5,PlayBack(vm-goodbye) exten=5,6,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Otis Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve Sorry to reply to my own post but for clarification, we had four queues. English sales, English support, Spanish sales, Spanish Support. At peek times, we would have 200-300 agents logged in and 600 or so callers. This was usually when our ads were running during Jerry Springer or Judge Judy. I think his two agent single queue would work just fine. Add Queuemetrics which is free (I believe) for five or less agents and then you can actually get some reporting on how your support role is handled. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Steve do you have an example of this... Otis Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Yes, but I have to be up very early in the morning and it is getting late. The answer priority will work for you in the meantime. If you want to investigate using real queues, let me know and I will help you set it up. Most of the stuff is on the Wiki but I will give you exact settings that should work on your setup. If you plan on growing or ever want to collect data on queues, then this is the way to go. Thanks, Steve OCOSA ListAcct wrote: Steve do you have an example of this... Otis Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Ok thanks. I will finish reading and see if I have any questions I will post and wait until you answer thank you! Otis Steve Totaro wrote: Yes, but I have to be up very early in the morning and it is getting late. The answer priority will work for you in the meantime. If you want to investigate using real queues, let me know and I will help you set it up. Most of the stuff is on the Wiki but I will give you exact settings that should work on your setup. If you plan on growing or ever want to collect data on queues, then this is the way to go. Thanks, Steve OCOSA ListAcct wrote: Steve do you have an example of this... Otis Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Yes. MOST of the time you should not use Answer, but in this specific case it may solve your issue. OCOSA ListAcct wrote: Eric so I should do this exten=5,1,Answer exten=5,2,StartMusicOnHold exten=5,3,Dial(SIP/supportSIP/support2,2,tr) exten=5,4,VoiceMail([EMAIL PROTECTED]) exten=5,5,PlayBack(vm-goodbye) exten=5,6,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Otis Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Steve Totaro wrote: Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve Sorry to reply to my own post but for clarification, we had four queues. English sales, English support, Spanish sales, Spanish Support. At peek times, we would have 200-300 agents logged in and 600 or so callers. This was usually when our ads were running during Jerry Springer or Judge Judy. I think his two agent single queue would work just fine. Add Queuemetrics which is free (I believe) for five or less agents and then you can actually get some reporting on how your support role is handled. In your situation it seems that queues work well for you. When you have dedicated agents answering calls full time queues work well. In non-call shops people forget to log out of the queue, are away from their desk often, and otherwise just screw up many of the assumptions that the Asterisk queue system makes. This is in addition to the learning curve. For a low number of calls and/or non-dedicated agents, a little bit of dialplan logic can do everything someone needs with something that is massively more flexible. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Steve / Eric When configuring the queue I tested works fine but one issue. My agent auto logs off after I am done with the call. I tried ignoring that option in agents.conf no luckAlso the below with the Answer line does not work either...still stays on and ring about 1:15 secs then goes to voicemail Otis Eric ManxPower Wieling wrote: Steve Totaro wrote: Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve Sorry to reply to my own post but for clarification, we had four queues. English sales, English support, Spanish sales, Spanish Support. At peek times, we would have 200-300 agents logged in and 600 or so callers. This was usually when our ads were running during Jerry Springer or Judge Judy. I think his two agent single queue would work just fine. Add Queuemetrics which is free (I believe) for five or less agents and then you can actually get some reporting on how your support role is handled. In your situation it seems that queues work well for you. When you have dedicated agents answering calls full time queues work well. In non-call shops people forget to log out of the queue, are away from their desk often, and otherwise just screw up many of the assumptions that the Asterisk queue system makes. This is in addition to the learning curve. For a low number of calls and/or non-dedicated agents, a little bit of dialplan logic can do everything someone needs with something that is massively more flexible. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. Couldn't one change the default timeout so that Dial() will ring for 2 seconds? Or will that have all kinds of other undesirable side effects? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 20min waiting time
OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. I wonder if this is your phone deciding it has been ringing for long enough and rejecting the call. Perhaps a NoOp(DialStatus is ${DIALSTATUS}) would shed light on this possibility? Trevor -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users