[asterisk-users] Realtime Queue - changing strategy to linear needs Asterisk restart
Hi group, We have realtime queue architecture on asterisk 1.8.7.0 I noticed that when we change strategy from any other to 'linear' it requires Asterisk restart take the change in effect. I have one realtime queue '1' with strategy set to 'ringall' and I change its strategy to 'linear'. Now when check on Asterisk CLI it shows me warning given below. demo*CLI queue show 1 1 has 0 calls (max 500) in 'ringall' strategy (0s holdtime, 0s talktime), W:1, C:0, A:0, SL:0.0% within 100s No Members No Callers [Nov 8 12:10:18] WARNING[4887]: app_queue.c:2034 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted. [Nov 8 12:10:18] WARNING[4887]: app_queue.c:2034 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted. This behaviour doesn't happen when strategy changed to other than 'linear'. So why is Asterisk restart needed for this change? TIA, --AM -- _ -- 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] Detecting Special Information Tone in Asterisk
Hi, Has anybody any idea about detecting Special Information Tone(SIT) when making utbound calls? http://en.wikipedia.org/wiki/Special_information_tone I googled for detecting SIT in Asterisk but couldn't find useful results. Thanks, --Sam -- _ -- 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] Asterisk 1.8.7.0- Incorrect information in Queue events-AMI
Opened the issue: ASTERISK-18707 On 10/12/11, Warren Selby wcse...@selbytech.com wrote: Just a guess at this point, but I'd say because you had two agents registered to the queue, but only one was available? If you dynamically logout the Unavailable agent, it should not show up in the QueueStatus response, however if you dynamically log him in then just shut off the phone, he'll still be listed as a member even if unavailable. Apparently, QueueSummary takes the available status into consideration when listing the number of logged in agents. You may be able to make a case that this behavior is inconsistent, and thus, a bug, but I could see it going either way. If you open a ticket on the issue, respond here with the issue id, I'd like to track it. Thanks, --Warren Selby, dCAP On Oct 11, 2011, at 11:39 PM, Asterisk Man theasterisk...@gmail.com wrote: Thanks Warren, I have been using X-lite for member and the system from where it is running was down at that time. My concern was, if 'Queuestatus' shows two members as logged in for the Queue then why not 'Queuesummary'? Any other pointer? --AM On Tue, Oct 11, 2011 at 8:34 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Oct 11, 2011 at 12:58 AM, Asterisk Man theasterisk...@gmail.com wrote: snip Event: QueueMember Queue: 1 Name: 3 Location: SIP/ Membership: dynamic Penalty: 2 CallsTaken: 0 LastCall: 0 Status: 5 Paused: 0 I would first troubleshoot why this Queue Member is showing up as Status: 5 (Unavailable). -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- 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] Asterisk 1.8.7.0- Incorrect information in Queue events-AMI
Thanks Warren, I have been using X-lite for member and the system from where it is running was down at that time. My concern was, if 'Queuestatus' shows two members as logged in for the Queue then why not 'Queuesummary'? Any other pointer? --AM On Tue, Oct 11, 2011 at 8:34 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Oct 11, 2011 at 12:58 AM, Asterisk Man theasterisk...@gmail.comwrote: snip Event: QueueMember Queue: 1 Name: 3 Location: SIP/ Membership: dynamic Penalty: 2 CallsTaken: 0 LastCall: 0 Status: 5 Paused: 0 I would first troubleshoot why this Queue Member is showing up as Status: 5 (Unavailable). -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- 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
[asterisk-users] Asterisk 1.8.7.0- Incorrect information in Queue events-AMI
Friends, I was just playing with couple of manager actions for Queue statistics on Asterisk 1.8.7.0 and found some inconsistency in information(I may be wrong somewhere interpreting the information!). Let me paste the outputs of my test for your reference. = [1] QueueStatus action: queuestatus queue: 1 Response: Success Message: Queue status will follow Event: QueueParams Queue: 1 Max: 0 Strategy: ringall Calls: 0 Holdtime: 2 TalkTime: 26 Completed: 20 Abandoned: 3 ServiceLevel: 0 ServicelevelPerf: 0.0 Weight: 0 Event: QueueMember Queue: 1 Name: 2 Location: SIP/1112 Membership: dynamic Penalty: 1 CallsTaken: 16 LastCall: 1318251310 Status: 1 Paused: 0 Event: QueueMember Queue: 1 Name: 3 Location: SIP/ Membership: dynamic Penalty: 2 CallsTaken: 0 LastCall: 0 Status: 5 Paused: 0 Event: QueueStatusComplete --- [2]QueueSummary action: queuesummary queue: 1 Response: Success Message: Queue summary will follow Event: QueueSummary Queue: 1 LoggedIn: 1 Available: 1 Callers: 0 HoldTime: 2 TalkTime: 26 LongestHoldTime: 0 Event: QueueSummaryComplete --- [3]CLI: Queue Show 10:35:00DEMO* queue show 1 1 has 0 calls (max unlimited) in 'ringall' strategy (2s holdtime, 26s talktime), W:0, C:20, A:3, SL:0.0% within 0s Members: 2 (SIP/1112) with penalty 1 (dynamic) (Not in use) has taken 16 calls (last was 58190 secs ago) 3 (SIP/) with penalty 2 (dynamic) (Unavailable) has taken no calls yet No Callers = You can see, =Response for action 'Queuestatus' shows 2 members logged in(Event: QueueMember) for Queue 1, whereas action 'Queuesummary' shows just 1(Event: QueueSummary Queue: 1 LoggedIn: 1 ...). Should I open the bug or am I missing something? Thanks, --AM -- _ -- 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] Asked to transmit frame type slin, while native formats is 0x8 (alaw)
Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have allowed only one codec(alaw). [Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) I also tried to yes/no option transcode_via_sln in asterisk.conf without any success. Any idea? Thanks, --AM -- _ -- 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] Asked to transmit frame type slin, while native formats is 0x8 (alaw)
Thanks for the response. I have disallow=all and allow=alaw in sip.conf for my SIP user. Any other idea? --AM On Tue, Jun 28, 2011 at 4:23 PM, Fellipe Paes fellipe...@hotmail.comwrote: Hello! In your sip.conf use alaw as your first codec option and see what happens. Best regards, Fellipe Paes -- Date: Tue, 28 Jun 2011 15:29:11 +0530 From: theasterisk...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw) Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have allowed only one codec(alaw). [Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) I also tried to yes/no option transcode_via_sln in asterisk.conf without any success. Any idea? Thanks, --AM -- _ -- 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
[asterisk-users] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.
Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest] exten = 1212,1,Noop(## Test mixmonitor with Queue ##) same = n,MixMonitor(testmixmonitorA.wav,W(4)) same = n,Queue(1,ct,,,50,,agntanserd) [macro-agntanserd] exten = s,1,Noop(## Agent answered the call. Record the call ##) same = n,MixMonitor(testmixmonitorB.wav,W(4)) I checked default path for recordings (/var/spool/asterisk/monitor) and it just shows a single recording for mixmonitor used before Queue()... [root@testmachine monitor]# ls testmixmonitorA.wav Following is the Asterisk CLI output... [May 5 17:26:34] -- Executing [1212@mixmonitortest:1] NoOp(SIP/31-001b, ## Test mixmonitor with Queue ##) in new stack [May 5 17:26:34] -- Executing [1212@mixmonitortest:2] MixMonitor(SIP/31-001b, testmixmonitorA.wav,W(4)) in new stack [May 5 17:26:34] -- Executing [1212@mixmonitortest:3] Queue(SIP/31-001b, 1,ct,,,50,,agntanserd) in new stack [May 5 17:26:34] == Begin MixMonitor Recording SIP/31-001b [May 5 17:26:34] -- Started music on hold, class 'default', on SIP/31-001b [May 5 17:26:34] WARNING[21215]: translate.c:162 framein: no samples for ulawtolin [May 5 17:26:34] == Using SIP RTP CoS mark 5 [May 5 17:26:34] -- SIP/1119-001c is ringing [May 5 17:26:40] -- SIP/1119-001c answered SIP/31-001b [May 5 17:26:40] -- Stopped music on hold on SIP/31-001b [May 5 17:26:40] -- Executing [s@macro-agntanserd:1] NoOp(SIP/1119-001c, ## Agent answered the call. Record the call ##) in new stack [May 5 17:26:40] -- Executing [s@macro-agntanserd:2] MixMonitor(SIP/1119-001c, testmixmonitorB.wav,W(4)) in new stack [May 5 17:26:40] == Begin MixMonitor Recording SIP/1119-001c [May 5 17:26:46] == End MixMonitor Recording SIP/1119-001c [May 5 17:26:46] == MixMonitor close filestream [May 5 17:26:46] == End MixMonitor Recording SIP/31-001b Any idead why is Asterisk not creating recording for Mixmonitor() application used in macro? Has anybody faced similar issue, or is a bug? Asterisk version- 1.8.3.2 I couldn't get chance to test on other Asterisk versions. Thanks in advance. --AM -- _ -- 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] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.
Thank you very much for your response and suggestion. I raised the question because in my project I don't want to record all the Queue calls. I just want to record calls connected with some specific members. --AM On Thu, May 5, 2011 at 11:10 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Thu, 2011-05-05 at 18:16 +0530, Asterisk Man wrote: Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest] exten = 1212,1,Noop(## Test mixmonitor with Queue ##) same = n,MixMonitor(testmixmonitorA.wav,W(4)) same = n,Queue(1,ct,,,50,,agntanserd) [macro-agntanserd] exten = s,1,Noop(## Agent answered the call. Record the call ##) same = n,MixMonitor(testmixmonitorB.wav,W(4)) I checked default path for recordings (/var/spool/asterisk/monitor) and it just shows a single recording for mixmonitor used before Queue()... [root@testmachine monitor]# ls testmixmonitorA.wav Following is the Asterisk CLI output... [May 5 17:26:34] -- Executing [1212@mixmonitortest:1] NoOp(SIP/31-001b, ## Test mixmonitor with Queue ##) in new stack [May 5 17:26:34] -- Executing [1212@mixmonitortest:2] MixMonitor(SIP/31-001b, testmixmonitorA.wav,W(4)) in new stack [May 5 17:26:34] -- Executing [1212@mixmonitortest:3] Queue(SIP/31-001b, 1,ct,,,50,,agntanserd) in new stack [May 5 17:26:34] == Begin MixMonitor Recording SIP/31-001b [May 5 17:26:34] -- Started music on hold, class 'default', on SIP/31-001b [May 5 17:26:34] WARNING[21215]: translate.c:162 framein: no samples for ulawtolin [May 5 17:26:34] == Using SIP RTP CoS mark 5 [May 5 17:26:34] -- SIP/1119-001c is ringing [May 5 17:26:40] -- SIP/1119-001c answered SIP/31-001b [May 5 17:26:40] -- Stopped music on hold on SIP/31-001b [May 5 17:26:40] -- Executing [s@macro-agntanserd:1] NoOp(SIP/1119-001c, ## Agent answered the call. Record the call ##) in new stack [May 5 17:26:40] -- Executing [s@macro-agntanserd:2] MixMonitor(SIP/1119-001c, testmixmonitorB.wav,W(4)) in new stack [May 5 17:26:40] == Begin MixMonitor Recording SIP/1119-001c [May 5 17:26:46] == End MixMonitor Recording SIP/1119-001c [May 5 17:26:46] == MixMonitor close filestream [May 5 17:26:46] == End MixMonitor Recording SIP/31-001b Any idead why is Asterisk not creating recording for Mixmonitor() application used in macro? Has anybody faced similar issue, or is a bug? Asterisk version- 1.8.3.2 I couldn't get chance to test on other Asterisk versions. What is wrong with the native Queue recording? Check queues.conf and make sure you have: monitor-type = MixMonitor monitor-format = gsm|wav|wav49 This will automatically record calls when the agent answers the call. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.
Further,
Before I start working on full project, I wanted to test the functionalities
to be implemented. So I wrote a small test dialplan to check whether I can
record a Queue call in Macro which gets executed on Member answer. My actual
macro would be like this...
[macro-agntanserd]
exten = s,1,Noop(## Agent answered the call. Record the call
##)
;-- Check whether to record a call or not --;
same =
n,Set(ARRAY(RECORDCALL,ONDEMAND)=${ODBC_CHECK_CALL_RECORDING(${MEMBERNAME})})
same = n,ExecIf($[${RECORDCALL} =
1]?MixMonitor(testmixmonitorB.wav,W(4)):Noop())
And I have a realtime Queue in which members are added/removed dynamically.
Any help or pointer will be appreciated.
Thanks,
--AM
On Fri, May 6, 2011 at 9:52 AM, Asterisk Man theasterisk...@gmail.comwrote:
Thank you very much for your response and suggestion.
I raised the question because in my project I don't want to record all the
Queue
calls. I just want to record calls connected with some specific members.
--AM
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[asterisk-users] How does wrandom strategy works with Queue?
Hi, wrandom strategy for Queue says...rings random interface, but uses the member's penalty as a weight when calculating their metric. So a member with penalty 0 will have a metric somewhere between 0 and 1000, and a member with penalty 1 will have a metric between 0 and 2000, and a member with penalty 2 will have a metric between 0 and 3000. Please note, if using this strategy, the member penalty is not the same as when using other queue strategies. It is ONLY used as a weight for calculating metric. Does it mean if a member with lower penalty is available, Queue will always send the call to that member? what if that member ignores the call(I am using X-lite as a member phone)? Should Queue ringback the same member or ring somebody else who has higher penalty and available? I tried this on Asterisk 1.8.0 and found different behaviors each time. First case it tried ringing the same lower penalty member no matter call was ignored or not. In another case it rang second member when lower penalty member ignored the call. Moreover,How does that matric get calculated and affect the behavior of ringing the member? Thanking you in advance... --AM -- _ -- 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] How does wrandom strategy works with Queue?
Thanks Jaron, I understood the point from your explanation. What should I do if I always want to ring a particular Queue member first whenever he is available? Yes, I can dial that member first before sending the call to Queue and achieve the result but just wanted to know views from others. Regards, --AM On Tue, Apr 26, 2011 at 2:13 PM, Jeroen Eeuwes jeroeneeu...@gmail.comwrote: Hi AM, I tried this on Asterisk 1.8.0 and found different behaviors each time. Isn't that part of the definition of random? If Asterisk would behave the same each time it wouldn't be random but predictable, I would say. AFAIK the metric just means that you get a higher or lower chance of being selected instead of being completely random. So instead of picking between person A or B choices it will choose between -let's say- 10 marbles. 3 of them are white and 7 of them are black. So black gets a higher chance of being selected. But it does not mean a white marble won't be selected. Best regards, Jeroen Eeuwes -- _ -- 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
[asterisk-users] Queue(): How to know Estimated wait time for caller in advance
Hi, Can we know the estimated wait time for a caller before sending him in a Queue? Asterisk 1.8 Thanks, --AM -- _ -- 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] Asterisk 1.8 Dimensioning.
Hi Group, Is there any information available for Asterisk 1.8 dimensioning? I googled but couldn't find helpful data for 1.8. I am trying to figure out hardware configuration for following features implemented in Asterisk 1.8? (1)100 SIP clients. (2)ACD (Around 15 realtime queues) (3)Call recording for all SIP clients. (4)4 port PRI (E1). There would be around 100 concurrent calls. (DAHDI2SIP,SIP2SIP,SIP2DAHDI) (5)IVR (6)Around 50 Mysql queries per call (through ODBC). (Remote Database) (7)MOH I can provide further information if missing something. Thanking you in advance. --AM -- _ -- 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(): how to Perform operations at the time of call sent to Queue member but not answered.
Hi Group, In Queue application, we have AGI,macro and gosub parameters that allow us to perform some operations when Queue member gets connected with caller. But it seems that right now there is no such mechanism (except CEL,AMI) for situation where we want some operations to be performed when call is sent to Queue Member but not answered yet (i.e. Queue Member interface is in ringing state). I know monitoring Channel events we can do this, but I wanted something in dialplan itself to get it done. Probably I will not be the only person asking for this future in Asterisk. PLease have your thoughts on this. Thanking you. --AM -- _ -- 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] Status of Queue Members
Probably this will help you... http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288901 Check the section 'Controlling when to join and leave a queue'. --AM On Thu, Mar 17, 2011 at 9:15 PM, Dan Journo d...@keshercommunications.comwrote: Hi, I'm trying to work out an issue with call queues. I need the calls that are in a queue to be kicked out if all members are unavailable (for example if all SIP members are having network problems). I tried leavewhenempty = yes but that only seems works when all queue members specifically log out of a queue. I've looked at autopause, but we need it to automatically un-pause once it comes back online. Any idea how I can do this? Preferably without using the AMI or AGI scripts, but if that's the only way, then i'll have to use that. Thanks Dan Kesher Communications (UK) Business Phone Systems http://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- 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
[asterisk-users] Answering machine detection for a second leg call generated by a call file.
Hi Group, I have following case scenario. Through call file, Asterisk makes a call to SIP extension. When Extension answers the call, Asterisk reads customer numbers (set in callfile) and calls them one by one untill one of the customers answeres the call. Here customer and SIP extension gets patched and talk to each other. Now if outgoing call is answered by Answering machine,I don't want asterisk to patch it up with SIP extension. Please suggest me how this can be achieved. Thanking you in advance. --AM -- _ -- 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] Answering machine detection for a second leg callgenerated by a call file.
Thanks buddy, But I think, AMD helps when I call customer first and then SIP extension. Any other suggestion! On Thu, Mar 17, 2011 at 6:44 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Asterisk Man *Sent:* Thursday, March 17, 2011 8:13 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Answering machine detection for a second leg callgenerated by a call file. Hi Group, I have following case scenario. Through call file, Asterisk makes a call to SIP extension. When Extension answers the call, Asterisk reads customer numbers (set in callfile) and calls them one by one untill one of the customers answeres the call. Here customer and SIP extension gets patched and talk to each other. Now if outgoing call is answered by Answering machine,I don't want asterisk to patch it up with SIP extension. Please suggest me how this can be achieved. Thanking you in advance. --AM May or may not help – google for “Asterisk AMD” -- _ -- 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] Answering machine detection for a second leg callgenerated by a call file.
This seems better. I will give it a try.
Thanks federico.
On Thu, Mar 17, 2011 at 11:10 PM, federico cabiddu
federico.cabi...@gmail.com wrote:
AMD is used mainly in scenarios like yours where an agent (the SIP
extension) is called, then an outbound call is generated and finally
the two legs are bridged. In your case you could call the Dial cmd
using the M option. The argument of M can be a macro like this simple
one:
exten = s,1,Background(short_silence)
exten = s,n,AMD()
exten = s,n,GotoIf($[${AMDSTATUS}=MACHINE]?mach:humn)
exten = s,n(humn),MacroExit
exten = s,n(mach),Set(MACRO_RESULT=CONTINUE)
So if an human is detected the legs will be bridged, if not the called
party will be hangup and the next number will be called.
The problem is, like previously said, the accuracy of the detection...
Best regards,
Federico
2011/3/17 Asterisk Man theasterisk...@gmail.com:
Thanks buddy,
But I think, AMD helps when I call customer first and then SIP extension.
Any other suggestion!
On Thu, Mar 17, 2011 at 6:44 PM, Danny Nicholas da...@debsinc.com
wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Man
Sent: Thursday, March 17, 2011 8:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Answering machine detection for a second leg
callgenerated by a call file.
Hi Group,
I have following case scenario.
Through call file, Asterisk makes a call to SIP extension. When
Extension
answers the call, Asterisk reads customer numbers (set in callfile) and
calls them one by one untill one of the customers answeres the call.
Here
customer and SIP extension gets patched and talk to each other.
Now if outgoing call is answered by Answering machine,I don't want
asterisk to patch it up with SIP extension. Please suggest me how this
can
be achieved.
Thanking you in advance.
--AM
May or may not help – google for “Asterisk AMD”
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[asterisk-users] How to know Caller's last position in Queue?
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a variable called ${QEORIGINALPOS} that gives us original
position of caller in Queue, but there doesn't seem to have something
similar for exit position.
Am I missing something?
Thanks,
--AsteriskMan
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Re: [asterisk-users] How to know Caller's last position in Queue?
Hi Hanif,
I indeed use 1.8 .0 but couldn't find the channel variable for caller's
last position in queue anywhere in documentation.
Would you please let me know the channel variable name?
Thanking you.
On Wed, Feb 16, 2011 at 4:40 PM, Faisal Hanif fai...@vopium.com wrote:
If you use Asterisk 1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Asterisk Man
*Sent:* Wednesday, February 16, 2011 3:06 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] How to know Caller's last position in Queue?
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a variable called ${QEORIGINALPOS} that gives us original
position of caller in Queue, but there doesn't seem to have something
similar for exit position.
Am I missing something?
Thanks,
--AsteriskMan
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[asterisk-users] Member penalty and Queue strategies
Hi Group, Does Queue application take member penalty into account when strategy is other than wrandom? If yes, What difference does it make in case of linear and rrmemory strategies? Thanking you, AsteriskMan -- _ -- 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] Call forwrading but call transfer back
Do you forward the call from SIP phone or Asterisk dialplan. If it is from SIP Phone, above solution will not work. Infact any solution will not work except your softphone supports call forwarding based on some filter parameters. --AsteriskMan On 1/5/11, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Tuesday, January 04, 2011 9:36 AM To: asterisk-users Subject: [asterisk-users] Call forwrading but call transfer back Hi All, I have weird requirement for call forwarding. I have forward all call from A to B extension because A is very busy and sometime not available so B is taking care of all forwarding call from A. but in some case B need to transfer call to A and in this case call coming back to B again because of forwarding enabled. How to get rid on this condition ? How could B can transfer call to A ? Thanks, Satish This is a job for ex-girlfriend logic. Set up your dialplan like this (A=1001, B=1002) Exten = 1001,verbose(extension A-1001 handling) Exten = 1001,n,dial(SIP/1002) Exten = 1001/1002,n,dial(SIP/1001) If you dial 1001 from anywhere except 1002, you get sent to 1002. If you dial 1001 from 1002, you get sent to 1001. -- _ -- 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 Member relationship and AstDB
Is my explanation of query not so clear??? I may go wrong while asking something, then I should be guided to put it in a different way! This is with due respect to all of you guys. As this is Asterisk Users group, anybody, trying to learn Asterisk, may ask even a silly question. And (s)he should expect atleast a single reply from digium guys if not from others! I am very well aware of the 'Openess nature' of Asterisk Project, but if we want to compete with other PBXs (Read proprietary PBXs; as per my study there doesn't seem any other opensource PBX project coming near to Asterisk... atleast for the moment), we should be little bit more responsive. Regards, On Mon, Dec 27, 2010 at 4:35 PM, Asterisk Man theasterisk...@gmail.comwrote: I need clarification on couple of issues of Realtime Queue. It seems that when Agents(Memebers) are added using AddQueueMember, Asterisk puts this Queue-Member relationship information into AstDB, So that on asterisk restart this can be preserved. My question is, why does asterisk not store call information for Queue (holdtime, talktime, W, C, A, SL%) in AstDB, So that it can also be retained on restart? Though Queue-Member relationship information is stored in AstDB, it still forgets number of calls taken by member on instance of asterisk restart. Thanking you, -AsteriskMan -- _ -- 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 Member relationship and AstDB
I need clarification on couple of issues of Realtime Queue. It seems that when Agents(Memebers) are added using AddQueueMember, Asterisk puts this Queue-Member relationship information into AstDB, So that on asterisk restart this can be preserved. My question is, why does asterisk not store call information for Queue (holdtime, talktime, W, C, A, SL%) in AstDB, So that it can also be retained on restart? Though Queue-Member relationship information is stored in AstDB, it still forgets number of calls taken by member on instance of asterisk restart. Thanking you, -AsteriskMan -- _ -- 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] Moving asterisk from one network to another.
A ton of thanks for useful information. Quite informative to keep in mind for somebody like me who is still learner! On Sun, Dec 26, 2010 at 5:21 PM, Sebastian s...@open-t.co.uk wrote: Hi, On 12/24/2010 12:37 PM, Asterisk Man wrote: Friends, Do we need to change any Asterisk configuration files (Or any file related to Asterisk for that matter) when we put Asterisk box from one network to another? I guess it really depends on your setup. If you have SIP trunks for example, they might contain the IP's of external and internal networks in the config files. Your firewall (if you have one) might contain IP's and network masks. It depends on how the box was originally setup. Sebastian It is assumed that DB is on the same box. Asterisk box has got Asterisk running in it with no issues. Probably, it should not complain. I tried to check for IP address in Asterisk files (using ‘find . | xargs grep 192.168.X.XX –sl’), but it seems that Asterisk does not store specific IP in file(s). Your thoughts on this if I m missing something. -AsteriskMan -- _ -- 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
[asterisk-users] Moving asterisk from one network to another.
Friends, Do we need to change any Asterisk configuration files (Or any file related to Asterisk for that matter) when we put Asterisk box from one network to another? It is assumed that DB is on the same box. Asterisk box has got Asterisk running in it with no issues. Probably, it should not complain. I tried to check for IP address in Asterisk files (using ‘find . | xargs grep 192.168.X.XX –sl’), but it seems that Asterisk does not store specific IP in file(s). Your thoughts on this if I m missing something. -AsteriskMan -- _ -- 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] Moving asterisk from one network to another.
Friends, Do we need to change any Asterisk configuration files (Or any file related to Asterisk for that matter) when we put Asterisk box from one network to another? It is assumed that DB is on the same box. Asterisk box has got Asterisk running in it with no issues. Probably, it should not complain. I tried to check for IP address in Asterisk files (using ‘find . | xargs grep 192.168.X.XX –sl’), but it seems that Asterisk does not store specific IP in file(s). Your thoughts on this if I m missing something. -AsteriskMan -- _ -- 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] Pass DTMF to IVR gateway through SIP phone conferencing.
Christian, Thanks for your response. In my case, I was asked to do it through SIP phone 3 way call functionality and not the Asterisk Meetme application. I wanted to know if any one had done something similar in past or not. I am short of PRI in my test environment and hence I can't test it practically. Well, I 'll try to implement it using Meetme. Regards, AsteriskMan -- _ -- 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] Pass DTMF to IVR gateway through SIP phone conferencing.
Will someone help/direct me find a way to implement this? Or you can suggest some other method. On Fri, Dec 17, 2010 at 12:44 PM, Asterisk Man theasterisk...@gmail.comwrote: Hi friends, I want to implement following scenario using Asterisk. Please suggest me whether it is possible or not. This is bit off Asterisk and more on SIP side. An Asterisk box with one Station(SIP channel) and PRI. Agent dials a PSTN number of customer from station through Asterisk PRI. Agent gets connected with customer. Agent puts customer on hold. Agent dials another PSTN number which is of IVR gateway. Agent now makes conference(Station facility) with customer and IVR gateway. Gateway plays an IVR asking customer to enter his customer id number. My question is, will DTMF get forwarded to IVR gateway? I am asked to implement this and not having PRI for the moment in my Asterisk box. Thanking you in advance. -AsteriskMan -- _ -- 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] Contradiction between 2 AMI actions QueueSummary and Queuestatus
Let me explain this with some more details. I have 2 members logged into Queue 'retailBanking' using AddQueueMember application on 2 different softphones. The softphones from which these 2 members were added, later unregistered from Asterisk. I then fired below mentioned AMI actions and observed the output. 'QueueSummary' didn't show any member logged into 'retailBanking', where as 'Queuestatus' did show members with 'QueueMember' event. Is this bug or intended behavior? Should I submit a bug report? On Fri, Dec 17, 2010 at 4:59 PM, Asterisk Man theasterisk...@gmail.comwrote: Asterisk Version: 1.8.0 Members are added through AddQueueMember in realtime Queues On Fri, Dec 17, 2010 at 4:52 PM, Asterisk Man theasterisk...@gmail.comwrote: Guys, Why is such contradiction between 2 AMI actions QueueSummary and Queuestatus? Look at LoggedIn of QueueSummary and Event: QueueMember. Also LongestHoldTime of QueueSummary does not give correct value. - Action: QueueSummary Queue: retailBanking Response: Success Message: Queue summary will follow Event: QueueSummary Queue: retailBanking LoggedIn: 0 Available: 0 Callers: 0 HoldTime: 22 TalkTime: 231 LongestHoldTime: 0 Event: QueueSummaryComplete - Action: Queuestatus Queue: retailBanking Response: Success Message: Queue status will follow Event: QueueParams Queue: retailBanking Max: 0 Strategy: rrmemory Calls: 0 Holdtime: 22 TalkTime: 231 Completed: 5 Abandoned: 4 ServiceLevel: 0 ServicelevelPerf: 0.0 Weight: 0 Event: QueueMember Queue: retailBanking Name: agent2 Location: SIP/1110 Membership: dynamic Penalty: 0 CallsTaken: 1 LastCall: 1292570332 Status: 5 Paused: 0 Event: QueueMember Queue: retailBanking Name: agent1 Location: SIP/ Membership: dynamic Penalty: 0 CallsTaken: 3 LastCall: 1292581231 Status: 5 Paused: 0 Event: QueueStatusComplete - --AsteriskMan -- _ -- 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] Pass DTMF to IVR gateway through SIP phone conferencing.
Hi friends, I want to implement following scenario using Asterisk. Please suggest me whether it is possible or not. This is bit off Asterisk and more on SIP side. An Asterisk box with one Station(SIP channel) and PRI. Agent dials a PSTN number of customer from station through Asterisk PRI. Agent gets connected with customer. Agent puts customer on hold. Agent dials another PSTN number which is of IVR gateway. Agent now makes conference(Station facility) with customer and IVR gateway. Gateway plays an IVR asking customer to enter his customer id number. My question is, will DTMF get forwarded to IVR gateway? I am asked to implement this and not having PRI for the moment in my Asterisk box. Thanking you in advance. -AsteriskMan -- _ -- 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] Contradiction between 2 AMI actions QueueSummary and Queuestatus
Guys, Why is such contradiction between 2 AMI actions QueueSummary and Queuestatus? Look at LoggedIn of QueueSummary and Event: QueueMember. Also LongestHoldTime of QueueSummary does not give correct value. - Action: QueueSummary Queue: retailBanking Response: Success Message: Queue summary will follow Event: QueueSummary Queue: retailBanking LoggedIn: 0 Available: 0 Callers: 0 HoldTime: 22 TalkTime: 231 LongestHoldTime: 0 Event: QueueSummaryComplete - Action: Queuestatus Queue: retailBanking Response: Success Message: Queue status will follow Event: QueueParams Queue: retailBanking Max: 0 Strategy: rrmemory Calls: 0 Holdtime: 22 TalkTime: 231 Completed: 5 Abandoned: 4 ServiceLevel: 0 ServicelevelPerf: 0.0 Weight: 0 Event: QueueMember Queue: retailBanking Name: agent2 Location: SIP/1110 Membership: dynamic Penalty: 0 CallsTaken: 1 LastCall: 1292570332 Status: 5 Paused: 0 Event: QueueMember Queue: retailBanking Name: agent1 Location: SIP/ Membership: dynamic Penalty: 0 CallsTaken: 3 LastCall: 1292581231 Status: 5 Paused: 0 Event: QueueStatusComplete - --AsteriskMan -- _ -- 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] Contradiction between 2 AMI actions QueueSummary and Queuestatus
Asterisk Version: 1.8.0 Members are added through AddQueueMember in realtime Queues On Fri, Dec 17, 2010 at 4:52 PM, Asterisk Man theasterisk...@gmail.comwrote: Guys, Why is such contradiction between 2 AMI actions QueueSummary and Queuestatus? Look at LoggedIn of QueueSummary and Event: QueueMember. Also LongestHoldTime of QueueSummary does not give correct value. - Action: QueueSummary Queue: retailBanking Response: Success Message: Queue summary will follow Event: QueueSummary Queue: retailBanking LoggedIn: 0 Available: 0 Callers: 0 HoldTime: 22 TalkTime: 231 LongestHoldTime: 0 Event: QueueSummaryComplete - Action: Queuestatus Queue: retailBanking Response: Success Message: Queue status will follow Event: QueueParams Queue: retailBanking Max: 0 Strategy: rrmemory Calls: 0 Holdtime: 22 TalkTime: 231 Completed: 5 Abandoned: 4 ServiceLevel: 0 ServicelevelPerf: 0.0 Weight: 0 Event: QueueMember Queue: retailBanking Name: agent2 Location: SIP/1110 Membership: dynamic Penalty: 0 CallsTaken: 1 LastCall: 1292570332 Status: 5 Paused: 0 Event: QueueMember Queue: retailBanking Name: agent1 Location: SIP/ Membership: dynamic Penalty: 0 CallsTaken: 3 LastCall: 1292581231 Status: 5 Paused: 0 Event: QueueStatusComplete - --AsteriskMan -- _ -- 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
