Re: [asterisk-users] Call Queues : linear strategy WITH priority
On Wednesday 12 Aug 2015, Jonas Kellens wrote: Hello I was wondering of it is possible to have Queue Agents with the same priority (penalty) but with a certain order ? So I have 20 Agents. Agent 1 till Agent 10 has penalty 1. Agent 11 till Agent 15 has penalty 2. (only contacted if 1 - 10 are busy) Agent 16 till Agent 20 has penalty 3. (only contacted if 1 - 10 and 11 - 15 are busy) Within the range of Agent 1 till Agent 10, can I have a certain order in these Agents in which they are rung ?? Like Agent 1 - Agent 5 - Agent 2 3 4 - Agent 6 - Agent 7 - Agent 8 9 10. What's wrong with giving agent 1 penalty 1; agent 5 penalty 2; agents 2, 3 and 4 penalty 3; agent 6 penalty 4, agent 7 penalty 5, and so forth? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 Queues : linear strategy WITH priority
On 12-08-15 16:31, A J Stiles wrote: On Wednesday 12 Aug 2015, Jonas Kellens wrote: Hello I was wondering of it is possible to have Queue Agents with the same priority (penalty) but with a certain order ? So I have 20 Agents. Agent 1 till Agent 10 has penalty 1. Agent 11 till Agent 15 has penalty 2. (only contacted if 1 - 10 are busy) Agent 16 till Agent 20 has penalty 3. (only contacted if 1 - 10 and 11 - 15 are busy) Within the range of Agent 1 till Agent 10, can I have a certain order in these Agents in which they are rung ?? Like Agent 1 - Agent 5 - Agent 2 3 4 - Agent 6 - Agent 7 - Agent 8 9 10. What's wrong with giving agent 1 penalty 1; agent 5 penalty 2; agents 2, 3 and 4 penalty 3; agent 6 penalty 4, agent 7 penalty 5, and so forth? By giving a different penalty to Agents 1 to 10, there is no order. With penalty, the Agent keeps on being contacted untill it takes the call. Many forget that this is how penalties work ! So in stead of going from Agent 1 to Agent 5 to Agent 2,3,4 it is very possible that Agent 5 keeps on ringing when Agent 1 is 'busy calling', in stead of going further to Agents 2,3,5. In your case, Agent 5 will be called over and over again untill it takes the call. Not exactly what I'm looking for. Kind regards, Jonas. -- _ -- 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] Call Queues : linear strategy WITH priority
Hello I was wondering of it is possible to have Queue Agents with the same priority (penalty) but with a certain order ? So I have 20 Agents. Agent 1 till Agent 10 has penalty 1. Agent 11 till Agent 15 has penalty 2. (only contacted if 1 - 10 are busy) Agent 16 till Agent 20 has penalty 3. (only contacted if 1 - 10 and 11 - 15 are busy) Within the range of Agent 1 till Agent 10, can I have a certain order in these Agents in which they are rung ?? Like Agent 1 - Agent 5 - Agent 2 3 4 - Agent 6 - Agent 7 - Agent 8 9 10. I guess I need 'linear' strategy, but will penalty option still work ? Thank you for your feedback. Kind regards. Jonas. -- _ -- 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 queues on load-balanced asterisks
Hi Pan Dhaval, In the past 8 weeks, we have delivered a load-balanced asterisks (1.4) based call center with our flexqueue application for icson.com. It has the below features, 1. 2 x asterisk 1.4 boxes, 1 x mysql db box and 1 x flexqueue box(the two are failover configured with heartbeat and custom script, and mysql master-slave replication between two svr), 2 x kamailio boxes(failover configured), 1 x file server boxes, 1 x app server , run freepbx queuemetrics. all 8 server are dell r310. 2. the gateway is one mx100 with 4 E1 lines plugged, the incoming calls go to kamailio2 , and routed to ast1/ast2 in round robin mode. 3. all agent phones registered to kamailio 1, and the extensions are still maintained with freepbx 4.On asterisks, all trunks with destination to pstn or agent phones, go to kamailio1; and incoming calls trunk from kamailio2. 5.admin also use freepbx to configure inbound routes, ivrs, announcements, timeconditions, and recordings , etc. the configuration files are generated on the fly for flexqueue when apply changes. Dialplans for inbound routes are also automatically generated and distributed to ast1 ast2, in these dialplan, fastagi application is installed as well to point to flexqueue. 6.flexqueue interprets the call flow configured on freepbx, and create the queues configured on freepbx, but it's shared among all asterisk boxes. 7.flexqueue interface with queuemetrics , and send all necessary queue logs to queuemetrics for complete reporting QA purpose. 8.flexqueue has a agent phone panel, and a supervisor monitoring management panel. Agent can logon his/her phone panel to have features like, incoming call popup, parking, Outbound dial, hold/unhold, transfer (cold/warm/to another queue), hangup, wrapup , pause/resume, etc. the supervisor can logon his/her monitoring management panel, to view realtime event-driven agent info, queue info, and calls on-going. Besides, supervisor can also listen to agents, barge agents' talk, and qc call records recordings quickly. 9.flexqueue provide web api for customer's CRM, which is asp.net based, to make agent can click-dial in their web crm application, and playback recordings to the agent's phone by clicking playback button beside crm communication records. The above system has been put into production from today, it's fully load-balanced asterisks based call queues or call centers. the gateways , the asterisk boxes can be added/removed any time. The fault asterisk box will be detected, and bypassed from routing destinations. I wish it's a good reference for your guys who want to create the same infrastructures. Best Regards, Thomas Liu - WShuttle Infotech Ltd. http://www.wshuttle.com / http://www.lookmypc.com http://www.vicidial.cn / http://www.call-center-software.com.cn Tel: +86 20 39230098 39230096 Mobile : +86 1390 3051 930 HK DID: +852 6950 0916, Macau DID: +853 6285 0645 Email: thomas@wshuttle.com MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly Yahoo Messenger: thomaslly Address: Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, Guangzhou Higher Education Mega Center, Guangzhou, Guangdong Province, China. Zip code: 510006 -- -Original Message- From: Thomas Liu [mailto:thomas@wshuttle.com] Sent: Wednesday, January 12, 2011 12:15 AM To: 'asterisk-users@lists.digium.com' Subject: RE: Call queues on load-balanced asterisks Hi Pan Dhaval, We have implemented a FastAGI based queue with Erlang for a inbound call center, and call this new application as FlexQueue. All calls distributed on multiple asterisk boxes go through and are controlled by that same remote fastagi server. It can routing calls to any destination, by any business rules. It don't rely on the db for agent/call status store query. It's event driven and dict based agent/call store query, with very good performance, and low cpu power consumption. I think for your requirement, app_queue could not fulfill that. Best Regards, Thomas Liu - WShuttle Infotech Ltd. http://www.wshuttle.com / http://www.lookmypc.com http://www.vicidial.cn / http://www.call-center-software.com.cn Tel: +86 20 39230098 39230096 Mobile : +86 1390 3051 930 HK DID: +852 6950 0916, Macau DID: +853 6285 0645 Email: thomas@wshuttle.com MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly Yahoo Messenger: thomaslly Address: Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, Guangzhou Higher Education Mega Center, Guangzhou, Guangdong Province, China. Zip code: 510006 -- Hello Dhaval (and others),
Re: [asterisk-users] Call queues on load-balanced asterisks
Hello Mr. Liu, I tried searching for more information about FlexQueue, where to download etc. Google linked to vicidial.cn, which appears in your signature, but that page is all in chinese, and I couldn't find any english link. Where can I get more information about it? Is it a commercial product? With kind regards, Pan B. Christensen Ibidium AS http://www.ibidium.no - Original Message - From: Thomas Liu thomas@wshuttle.com To: asterisk-users@lists.digium.com Sent: Tuesday, January 11, 2011 5:14 PM Subject: Re: [asterisk-users] Call queues on load-balanced asterisks Hi Pan Dhaval, We have implemented a FastAGI based queue with Erlang for a inbound call center, and call this new application as FlexQueue. All calls distributed on multiple asterisk boxes go through and are controlled by that same remote fastagi server. It can routing calls to any destination, by any business rules. It don't rely on the db for agent/call status store query. It's event driven and dict based agent/call store query, with very good performance, and low cpu power consumption. I think for your requirement, app_queue could not fulfill that. Best Regards, Thomas Liu - WShuttle Infotech Ltd. http://www.wshuttle.com / http://www.lookmypc.com http://www.vicidial.cn / http://www.call-center-software.com.cn Tel: +86 20 39230098 39230096 Mobile : +86 1390 3051 930 HK DID: +852 6950 0916, Macau DID: +853 6285 0645 Email: thomas@wshuttle.com MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly Yahoo Messenger: thomaslly Address: Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, Guangzhou Higher Education Mega Center, Guangzhou, Guangdong Province, China. Zip code: 510006 -- Hello Dhaval (and others), As far as I can tell, realtime queue will not solve my problem. I can statically define the same queue with the same members on two machines as well. I was planning to use realtime anyway. The issue is the actual queueing of the incoming calls. Let?s say I define the queue IT-support with members Local/100 and Local/101 on both machines. The first call comes in and is distributed by Kamailio to Asterisk A, and answered by 100. The next call comes in to Asterisk B, and is answered by 101. At this point, both members are busy. Call 3 now comes in and is sent to Asterisk A, where it waits for a free member. Call 4 comes in and is also sent to Asterisk A, as is Call 5. Then call 6 is sent to Asterisk B. At this point 100 finishes his call and becomes free. Which call is delivered to 100? As far as I can tell, that?s a 50/50 chance between call 3 and call 6. This is not correct behaviour! Call 6 should wait until calls 3, 4 and 5 (from the other server) have all been delivered. In the example above: When call 3 comes in, Asterisk A may even try to deliver it to 101, who gets call waiting indication. He will now have two simultaneous calls from the same queue! I have not found any way to share information about calls waiting in the queue, wait times, member states and so on between the two servers. Unless you guys know of a way, I think I'm going to have to ask the customer to change their design to master-slave (with failover) instead of load-balanced. With kind regards, Pan Hello Pan, You can user DB for this just make real time configuration of Queue and make all asterisk server connected to Same DB if more load then use replication for different server on DB, also So that Quque name should be same for all server and asterisk can call same agent. you didnot mentioned that which purpose youwere use queue other wise i can give answer in better way. regards Dhaval On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen pan at ibidium.no wrote: Hello, I have been asked to implement the following design: Load-balanced Kamailio servers handling registrations and routing. Load-balanced asterisk feature servers handling voicemail and other things Kamailio cannot do. Plus several load-balanced gateways, but they are not relevant to my question. All this is working fine. I've now been asked to start implementing calling queues, and my question is this: How can I implement the same queue on multiple Asterisk servers? Let's say that 10 people call the same queue. These calls would then currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make Asterisk A respect the 5 people queued on the other server and vice versa? Will the customer need to change their design to make the feature servers master-slave with failover instead of load-balanced? Mvh Pan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
Re: [asterisk-users] Call queues on load-balanced asterisks
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. Christensen Sent: Tuesday, January 11, 2011 5:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call queues on load-balanced asterisks Hello Dhaval (and others), As far as I can tell, realtime queue will not solve my problem. I can statically define the same queue with the same members on two machines as well. I was planning to use realtime anyway. The issue is the actual queueing of the incoming calls. Let's say I define the queue IT-support with members Local/100 and Local/101 on both machines. The first call comes in and is distributed by Kamailio to Asterisk A, and answered by 100. The next call comes in to Asterisk B, and is answered by 101. At this point, both members are busy. Call 3 now comes in and is sent to Asterisk A, where it waits for a free member. Call 4 comes in and is also sent to Asterisk A, as is Call 5. Then call 6 is sent to Asterisk B. At this point 100 finishes his call and becomes free. Which call is delivered to 100? As far as I can tell, that's a 50/50 chance between call 3 and call 6. This is not correct behaviour! Call 6 should wait until calls 3, 4 and 5 (from the other server) have all been delivered. In the example above: When call 3 comes in, Asterisk A may even try to deliver it to 101, who gets call waiting indication. He will now have two simultaneous calls from the same queue! I have not found any way to share information about calls waiting in the queue, wait times, member states and so on between the two servers. Unless you guys know of a way, I think I'm going to have to ask the customer to change their design to master-slave (with failover) instead of load-balanced. With kind regards, Pan IMO your best solution to this is going to be using a database and AGI query to keep a quasi-real (delayed by a few ms/sec) picture of the queue activity. If you kept a database on both machines and ran an AGI with each incoming call to query queue usage on both machines or better yet, query the queue on the remote machine and spawn a short local call to keep that agent busy on the native machine, that would solve this issue. Let's say that a typical agent interaction occurs in 60 second chunks. Call 1 comes in to machine 1 and is answered by agent 100 as you said. Call 2 comes into machine 2 and is answered by 101. When Call 3 comes in, it sees 101 and 102 as busy on both machines. You can do this, but isn't this really a Kamailio issue? -- _ -- 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 queues on load-balanced asterisks
Hi Pan Dhaval, We have implemented a FastAGI based queue with Erlang for a inbound call center, and call this new application as FlexQueue. All calls distributed on multiple asterisk boxes go through and are controlled by that same remote fastagi server. It can routing calls to any destination, by any business rules. It don't rely on the db for agent/call status store query. It's event driven and dict based agent/call store query, with very good performance, and low cpu power consumption. I think for your requirement, app_queue could not fulfill that. Best Regards, Thomas Liu - WShuttle Infotech Ltd. http://www.wshuttle.com / http://www.lookmypc.com http://www.vicidial.cn / http://www.call-center-software.com.cn Tel: +86 20 39230098 39230096 Mobile : +86 1390 3051 930 HK DID: +852 6950 0916, Macau DID: +853 6285 0645 Email: thomas@wshuttle.com MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly Yahoo Messenger: thomaslly Address: Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, Guangzhou Higher Education Mega Center, Guangzhou, Guangdong Province, China. Zip code: 510006 -- Hello Dhaval (and others), As far as I can tell, realtime queue will not solve my problem. I can statically define the same queue with the same members on two machines as well. I was planning to use realtime anyway. The issue is the actual queueing of the incoming calls. Let?s say I define the queue IT-support with members Local/100 and Local/101 on both machines. The first call comes in and is distributed by Kamailio to Asterisk A, and answered by 100. The next call comes in to Asterisk B, and is answered by 101. At this point, both members are busy. Call 3 now comes in and is sent to Asterisk A, where it waits for a free member. Call 4 comes in and is also sent to Asterisk A, as is Call 5. Then call 6 is sent to Asterisk B. At this point 100 finishes his call and becomes free. Which call is delivered to 100? As far as I can tell, that?s a 50/50 chance between call 3 and call 6. This is not correct behaviour! Call 6 should wait until calls 3, 4 and 5 (from the other server) have all been delivered. In the example above: When call 3 comes in, Asterisk A may even try to deliver it to 101, who gets call waiting indication. He will now have two simultaneous calls from the same queue! I have not found any way to share information about calls waiting in the queue, wait times, member states and so on between the two servers. Unless you guys know of a way, I think I'm going to have to ask the customer to change their design to master-slave (with failover) instead of load-balanced. With kind regards, Pan Hello Pan, You can user DB for this just make real time configuration of Queue and make all asterisk server connected to Same DB if more load then use replication for different server on DB, also So that Quque name should be same for all server and asterisk can call same agent. you didnot mentioned that which purpose youwere use queue other wise i can give answer in better way. regards Dhaval On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen pan at ibidium.no wrote: Hello, I have been asked to implement the following design: Load-balanced Kamailio servers handling registrations and routing. Load-balanced asterisk feature servers handling voicemail and other things Kamailio cannot do. Plus several load-balanced gateways, but they are not relevant to my question. All this is working fine. I've now been asked to start implementing calling queues, and my question is this: How can I implement the same queue on multiple Asterisk servers? Let's say that 10 people call the same queue. These calls would then currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make Asterisk A respect the 5 people queued on the other server and vice versa? Will the customer need to change their design to make the feature servers master-slave with failover instead of load-balanced? Mvh Pan -- _ -- 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 queues on load-balanced asterisks
Hello Pan, You can user DB for this just make real time configuration of Queue and make all asterisk server connected to Same DB if more load then use replication for different server on DB, also So that Quque name should be same for all server and asterisk can call same agent. you didnot mentioned that which purpose youwere use queue other wise i can give answer in better way. regards Dhaval On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen p...@ibidium.no wrote: Hello, I have been asked to implement the following design: Load-balanced Kamailio servers handling registrations and routing. Load-balanced asterisk feature servers handling voicemail and other things Kamailio cannot do. Plus several load-balanced gateways, but they are not relevant to my question. All this is working fine. I've now been asked to start implementing calling queues, and my question is this: How can I implement the same queue on multiple Asterisk servers? Let's say that 10 people call the same queue. These calls would then currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make Asterisk A respect the 5 people queued on the other server and vice versa? Will the customer need to change their design to make the feature servers master-slave with failover instead of load-balanced? Mvh Pan -- _ -- 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] Call queues on load-balanced asterisks
Hello, I have been asked to implement the following design: Load-balanced Kamailio servers handling registrations and routing. Load-balanced asterisk feature servers handling voicemail and other things Kamailio cannot do. Plus several load-balanced gateways, but they are not relevant to my question. All this is working fine. I've now been asked to start implementing calling queues, and my question is this: How can I implement the same queue on multiple Asterisk servers? Let's say that 10 people call the same queue. These calls would then currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make Asterisk A respect the 5 people queued on the other server and vice versa? Will the customer need to change their design to make the feature servers master-slave with failover instead of load-balanced? Mvh Pan-- _ -- 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] Call queues - issues, can't make it work.
Hello there
I have been struggling with queues, because i think this is the right module
for our business.
My main goal, is when we receive external calls, the receptionist should be
able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Extension 301 and 302.
I have a test setup now which I thought should look like this:
When a external call come to my external number (67209611) this will ring for 5
seconds, and then transferred to queue teknisk
And I thought that internal phonex/extensions 301 and 302 would ring.
But, when I ring the external number, it just rings...and rings...until it
hang-ups.
CLI output shows that the commands are running, but maybe the wrong way, are
the queue command routed to my sip provider?
Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in technician department, which I am trying
to route the queue to with the ringall argument.
This happens:
Reloading MGCP
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [4767209...@internal:1]
NoOp(SIP/odin.service.ipallover.net-00d1, ) in new stack
-- Executing [4767209...@internal:2]
Verbose(SIP/odin.service.ipallover.net-00d1, Callerid num 90015103) in
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3]
Dial(SIP/odin.service.ipallover.net-00d1, SIP/301,5) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4]
Queue(SIP/odin.service.ipallover.net-00d1, teknisk) in new stack
-- Started music on hold, class 'default', on channel
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
-- SIP/odin.service.ipallover.net-00d1 Playing 'queue-youarenext.gsm'
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue
position (which was 1)
-- SIP/odin.service.ipallover.net-00d1 Playing 'queue-thankyou.gsm'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
== Spawn extension (internal, 4767209611, 4) exited non-zero on
'SIP/odin.service.ipallover.net-00d1'
asterisk*CLI
---
Agents.conf is default and i have two extensions/agents
agent = 301,301
agent = 302,302
--
[r...@asterisk asterisk]# more queues.conf
[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes
member = Agent/301
member = Agent/302
-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped
;exten 301
exten = 4767209611,1,NoOp();
exten = 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten = 4767209611,n,Dial(SIP/301,5);
exten = 4767209600,n,Queue(teknisk);
exten = 4767209611,n,Voicemail(301); ;Added 06.Mai.10-Aksel
Could someone please help me in the right direction here?
Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner
Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.nomailto:ak...@abacus-it.no
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
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Re: [asterisk-users] Call queues - issues, can't make it work.
when you add an agent to a queue the agent should log in try adding
member=SIP/301member=SIP/302instead of agent directives.this will ring both
phones.. from your output it doesn't seem to be ringing the agents at all.
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
From: ak...@abacus-it.no
To: asterisk-users@lists.digium.com
Date: Mon, 14 Jun 2010 13:41:20 +0200
Subject: [asterisk-users] Call queues - issues, can't make it work.
Hello there
I have been struggling with queues, because
i think this is the right module for our business.
My main goal, is when we receive external
calls, the receptionist should be able to transfer the call to us
Technicians, and I am trying to add 2
extensions to a queue name [teknisk]
Extension 301 and 302.
I have a test setup now which I thought
should look like this:
When a external call come to my external
number (67209611) this will ring for 5 seconds, and then transferred to queue
“teknisk”
And I thought that internal
phonex/extensions 301 and 302 would ring.
But, when I ring the external number, it
just rings…and rings…until it hang-ups.
CLI output shows that the commands are
running, but maybe the wrong way, are the queue command routed to my sip
provider?
Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in
technician department, which I am trying to route the queue to with the ringall
argument.
This happens:
Reloading MGCP
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing
[4767209...@internal:1]
NoOp(SIP/odin.service.ipallover.net-00d1, ) in new
stack
-- Executing
[4767209...@internal:2]
Verbose(SIP/odin.service.ipallover.net-00d1, Callerid num
90015103) in new stack
Callerid num 90015103
-- Executing
[4767209...@internal:3]
Dial(SIP/odin.service.ipallover.net-00d1,
SIP/301,5) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is
ringing
-- Nobody picked up in
5000 ms
-- Executing
[4767209...@internal:4]
Queue(SIP/odin.service.ipallover.net-00d1, teknisk)
in new stack
-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1
--
SIP/odin.service.ipallover.net-00d1 Playing 'queue-youarenext.gsm'
(language 'en')
-- Told
SIP/odin.service.ipallover.net-00d1 in teknisk their queue position (which
was 1)
--
SIP/odin.service.ipallover.net-00d1 Playing 'queue-thankyou.gsm'
(language 'en')
-- Started music on
hold, class 'default', on channel 'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold
on SIP/odin.service.ipallover.net-00d1
== Spawn extension (internal,
4767209611, 4) exited non-zero on 'SIP/odin.service.ipallover.net-00d1'
asterisk*CLI
---
Agents.conf is default and i have two
extensions/agents
agent = 301,301
agent = 302,302
--
[r...@asterisk asterisk]# more queues.conf
[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes
member = Agent/301
member = Agent/302
-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped
;exten 301
exten = 4767209611,1,NoOp();
exten = 4767209611,n,Verbose(Callerid
num ${CALLERID(num)});
exten = 4767209611,n,Dial(SIP/301,5);
exten = 4767209600,n,Queue(teknisk);
exten =
4767209611,n,Voicemail(301);
;Added 06.Mai.10-Aksel
Could someone please help me in the right
direction here?
Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner
Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no
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Re: [asterisk-users] Call queues - issues, can't make it work.
Thank You Tarek!
That was the case, and i saw now i had a typo in the extension further down,
but, you solved it.
Now I faced a couple of other problems, alle the announcements and MOH didn’t
play, the settings are default.
Maybe i'll figure it out.
Thank you
Regards
Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Tarek Sawah
Sendt: 14. juni 2010 15:00
Til: Asterisk Users
Emne: Re: [asterisk-users] Call queues - issues, can't make it work.
when you add an agent to a queue the agent should log in
try adding
member=SIP/301
member=SIP/302
instead of agent directives.
this will ring both phones.. from your output it doesn't seem to be ringing the
agents at all.
-- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347
562 2308
From: ak...@abacus-it.no
To: asterisk-users@lists.digium.com
Date: Mon, 14 Jun 2010 13:41:20 +0200
Subject: [asterisk-users] Call queues - issues, can't make it work.
Hello there
I have been struggling with queues, because i think this is the right module
for our business.
My main goal, is when we receive external calls, the receptionist should be
able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Extension 301 and 302.
I have a test setup now which I thought should look like this:
When a external call come to my external number (67209611) this will ring for 5
seconds, and then transferred to queue “teknisk”
And I thought that internal phonex/extensions 301 and 302 would ring.
But, when I ring the external number, it just rings…and rings…until it hang-ups.
CLI output shows that the commands are running, but maybe the wrong way, are
the queue command routed to my sip provider?
Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in technician department, which I am trying
to route the queue to with the ringall argument.
This happens:
Reloading MGCP
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [4767209...@internal:1]
NoOp(SIP/odin.service.ipallover.net-00d1, ) in new stack
-- Executing [4767209...@internal:2]
Verbose(SIP/odin.service.ipallover.net-00d1, Callerid num 90015103) in
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3]
Dial(SIP/odin.service.ipallover.net-00d1, SIP/301,5) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4]
Queue(SIP/odin.service.ipallover.net-00d1, teknisk) in new stack
-- Started music on hold, class 'default', on channel
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
-- SIP/odin.service.ipallover.net-00d1 Playing 'queue-youarenext.gsm'
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue
position (which was 1)
-- SIP/odin.service.ipallover.net-00d1 Playing 'queue-thankyou.gsm'
(language 'en')
-- Started music on hold, class 'default', on channel
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
== Spawn extension (internal, 4767209611, 4) exited non-zero on
'SIP/odin.service.ipallover.net-00d1'
asterisk*CLI
---
Agents.conf is default and i have two extensions/agents
agent = 301,301
agent = 302,302
--
[r...@asterisk asterisk]# more queues.conf
[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes
member = Agent/301
member = Agent/302
-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped
;exten 301
exten = 4767209611,1,NoOp();
exten = 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten = 4767209611,n,Dial(SIP/301,5);
exten = 4767209600,n,Queue(teknisk);
exten = 4767209611,n,Voicemail(301); ;Added 06.Mai.10-Aksel
Could someone please help me in the right direction here?
Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner
Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.nomailto:ak...@abacus-it.no
The New Busy think 9
Re: [asterisk-users] Call queues
Floyd wrote:
Hi everyone:
I've searching for a while and haven't found what i
need.
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to
be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
congestion when some of them(more than two!!) want
to
call outside, that is why i want to be able to put
those outgoing calls in a queue. For example if i
want
to call someone in the pstn and the fxo port is
already in use, i want to be placed in a queue and
when the channel is free my call is routed to the
aproppiated destination. As far as i have read the
queues are not for this kind of stuffs, there are
just agents or extensions that attend the calls in
the
queue and nothing more. am i wrong???
Any help will be useful.
thanks in advance!!
Hi,
first of all i would like to thanks to C. Chad Wallace
Noah and Rob Schall. I just solve the problem of the
outgoing call queue. Following the instructions from
Chad
i did something like this in my extensions.conf:
exten = _9XXX,1,Answer
exten = _9XXX,2,Set(_number=${EXTEN:1})
exten = _9XXX,3,Wait(2)
exten = _9XXX,4,NoOP(${number})
exten = _9XXX,5,Queue(Myqueue)
exten = _9XXX,6,Hangup
I also have a context like this
[outbound]
exten = 1,1,NoOP(${number})
exten = 1,2,Dial(Zap/G2/${number},30,t)
exten = 2,1,NoOP(${number})
exten = 2,2,Dial(Zap/G2/${number},30,t)
Finally in queues.conf
[myqueue]
member = Local/[EMAIL PROTECTED]/n
member = Local/[EMAIL PROTECTED]/n
And it worked perfect!!
I have my outgoing calls routed ok and the variable
travells throgh the queue without problems..
thanks!!
Eve
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[asterisk-users] Call Queues
Hi everyone: I've searching for a while and haven't found what i need. The thing is that i have a tdm422p with the two fxo ports connected to the pstn. I want my sip users to be able to call other numbers(any number) in the pstn through my zap fxo channels. I have a big number of sip users so as you can imagine there will be congestion when some of them(more than two!!) want to call outside, that is why i want to be able to put those outgoing calls in a queue. For example if i want to call someone in the pstn and the fxo port is already in use, i want to be placed in a queue and when the channel is free my call is routed to the aproppiated destination. As far as i have read the queues are not for this kind of stuffs, there are just agents or extensions that attend the calls in the queue and nothing more. am i wrong??? Any help will be useful. thanks in advance!! eve __ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.espanol.yahoo.com/ ___ --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] Call Queues
Hi Eve - The thing is that i have a tdm422p with the two fxo ports connected to the pstn. I want my sip users to be able to call other numbers(any number) in the pstn through my zap fxo channels. I have a big number of sip users so as you can imagine there will be congestion when some of them(more than two!!) want to call outside, that is why i want to be able to put those outgoing calls in a queue. For example if i want to call someone in the pstn and the fxo port is already in use, i want to be placed in a queue and when the channel is free my call is routed to the aproppiated destination. As far as i have read the queues are not for this kind of stuffs, there are just agents or extensions that attend the calls in the queue and nothing more. am i wrong??? I think your suspicions may be correct. You could add your ZAP channels as members in queues.conf, maybe something like this: members = ZAP/1, and then use queue() on your outbound extensions. The problem is how will your agents, in this case your ZAP trunks, know to pick up the line when they are not busy. You'd have to get these lines to somehow go offhook if they're not already busy. Maybe you can do this with an AGI script. I don't know, I've never tried to artificially control hook status. Personally, I'd probably just skip the whole queue idea and get some cheap SIP or IAX trunks and fall back to them when the ZAP lines are busy. - Noah ___ --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] Call Queues
Noah Miller wrote: Hi Eve - The thing is that i have a tdm422p with the two fxo ports connected to the pstn. I want my sip users to be able to call other numbers(any number) in the pstn through my zap fxo channels. I have a big number of sip users so as you can imagine there will be congestion when some of them(more than two!!) want to call outside, that is why i want to be able to put those outgoing calls in a queue. For example if i want to call someone in the pstn and the fxo port is already in use, i want to be placed in a queue and when the channel is free my call is routed to the aproppiated destination. As far as i have read the queues are not for this kind of stuffs, there are just agents or extensions that attend the calls in the queue and nothing more. am i wrong??? I think your suspicions may be correct. You could add your ZAP channels as members in queues.conf, maybe something like this: members = ZAP/1, and then use queue() on your outbound extensions. The problem is how will your agents, in this case your ZAP trunks, know to pick up the line when they are not busy. You'd have to get these lines to somehow go offhook if they're not already busy. Maybe you can do this with an AGI script. I don't know, I've never tried to artificially control hook status. Personally, I'd probably just skip the whole queue idea and get some cheap SIP or IAX trunks and fall back to them when the ZAP lines are busy. - Noah ___ --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 Possibly do a combination of things. Check if those zap chans are in use/busy. If they care, then create a call file using a script. I haven't played too much with it, so I don't know if those will queue until they can complete or if it will just error and delete itself. If you really are determined, you might even be able to route all requests to a script. Then have it check if there are any open lines... if so, create the call file... if not, then put it in a queue (in python, etc... not an asterisk queue), and try again in a min and see if a channel has opened up. Disclaimer - I have no idea if this idea will work. :) ___ --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] Call Queues
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Floyd wrote: Hi everyone: I've searching for a while and haven't found what i need. The thing is that i have a tdm422p with the two fxo ports connected to the pstn. I want my sip users to be able to call other numbers(any number) in the pstn through my zap fxo channels. I have a big number of sip users so as you can imagine there will be congestion when some of them(more than two!!) want to call outside, that is why i want to be able to put those outgoing calls in a queue. For example if i want to call someone in the pstn and the fxo port is already in use, i want to be placed in a queue and when the channel is free my call is routed to the aproppiated destination. As far as i have read the queues are not for this kind of stuffs, there are just agents or extensions that attend the calls in the queue and nothing more. am i wrong??? Any help will be useful. thanks in advance!! You could probably do this using the Local channel. You'd create a context, say outbound, to take calls from the queue and connect them to a Zap channel, with 2 extensions in that context--one for each channel. Then you add each of those extensions as members of the queue: member = Local/[EMAIL PROTECTED]/n member = Local/[EMAIL PROTECTED]/n Make sure your dialplan in outbound returns Busy if the Zap channel is busy. The tricky part would be passing the dialed number through... But if you set an inheriting channel var, it should go through the queue and into the Local channel to your outbound extension. Sorry I don't have any code for you... I haven't done it yet; I'm just putting the idea out there. Hope this helps! Good luck. - -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGjWCTKeSNHCYiCKARAtdgAKCVUs6OF2KIpjbpwQFrwr2E4NatVACfWh6I 9XwYqQ7cc5gwVznybIglBGs= =miEL -END PGP SIGNATURE- ___ --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] Call queues, agents with DND status set.
there should be a way in agents.conf to autologoff agents after a while the do not answer the phone. l. In data Sat, 03 Dec 2005 23:48:05 +0100, Vladimir S. Blazhkun [EMAIL PROTECTED] ha scritto: -- Called 1101 -- Agent/1101 is ringing -- Got SIP response 480 Temporarily Unavailable back from x.x.x.x -- SIP/1101-9b08 is circuit-busy Is it possible to force logoff such agents? -- Assum est, versa et manduca. ___ --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] Call queues, agents with DND status set.
-- Called 1101 -- Agent/1101 is ringing -- Got SIP response 480 Temporarily Unavailable back from x.x.x.x -- SIP/1101-9b08 is circuit-busy Is it possible to force logoff such agents? -- Vladimir S. Blazhkun, Personal Communications Systems, LLC. Leading IP NCC Specialist, Work phone: +7 095 7847617. JNCIA-M #773, JNCIS-M #1100. ___ --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] Call Queues
Hi, I do have two questions regarding call queues: 1) How can I reach that waiting calls are also removed on removing the last agent listening to the queue. All I found is the switch to prevent new calls enter the queue after the last agent left. 2) Currently my queue does ring the agent after playing the you are first. How can I have the phone start ringing while the message is played? Elmar ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queues and Agent Call Logs/Wrapup logs
Now that we have a well functioning Asterisk system that queues our calls and distributes them to our CSRs, I would like to implement a better system for our agents to keep a log of all of their calls, which we currently do using MS Word. (As you would expect, this is a less than ideal solution!) I am looking for a simple program that will allow our agents to enter notes on each call that they take and save it to a database. Basically, The call would come in, and I would use something like astGUIClient or IPSwitchboard to perform a screenpop to a web page or program that would let the agent type in notes re: their call. Something along the lines of: Sally Jones called re: Account #1234 and wanted to know what her balance was and if she could make a payment by credit card. I took her information, charged the card and faxed her the receipt. Ideally, the note would be saved with the CDR data so I could search by extension, date, etc and find all relevant entries. Earlier in the month something similar was mentioned, and the poster mentioned searching Google, but I have yet to find an appropriate solution, and before I go and try to reinvent this wheel, I thought I would ask the list members if they have implemented something similar. Please let me know if you have any suggestions. Thank you for your consideration, Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call queues bug?
Please has anyone experienced a bug with queues in Asterisk? No matter what settings my queue always thinks it got agents available in it. Plus if I take all the members out, then calls don't join the queue even though I've specified join-empty. Any advice? Neil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call queues
Hi, I'm want to do something slighty different with call queues than the config allows... I wish to have things work in an 'overflowish' manner. Ie, it works just like 'roundrobin', where it rings on one phone, no answer, rings on the next etc etc, except I want it to keep ringing on all phones that have rung so far. For example, call comes in, phone 1 rings, in 15 seconds, phone 2 starts to ring while phone 1 keeps rining. in a further 15 seconds, phone 3 starts to ring while 1 and 2 are still ringing etc etc. I'd code it in myself, but I'm not a C coder :) Does anyone have any solutions for this? The only way I personally could potentially do something like this is with some dialplan magic and an AGI script perhaps. Cheers, -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call queues problem
Hi there i just setup a asterisk box with autoattendant and call queues, but it seems that when one of the agents is busy all the new calls will stay on hold until The agent hangs up then all phone will ring [aftersales] musiconhold = default timeout = 15 retry = 5 maxlen = 0 member = sip/131 member = sip/132 member = sip/133 member = sip/134 this is the queue hope someone can help me here ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call queues problem
I think you have missed something with your agents.conf and with the member lines in queues.conf. This works for us: In queues.conf: [gws-wartefeld] music = default strategy = ringall context = queue-out timeout = 15 wrapuptime=10 announce-frequency = 0 announce-holdtime = no queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou ; (Thank you for your patience.) queue-lessthan = queue-less-than; (less than) queue-reporthold = queue-reporthold ; (Hold time) joinempty = no member = Agent/6301 member = Agent/6302 member = Agent/6303 member = Agent/6304 member = Agent/6305 In agents.conf: [general] persistentagents=yes [agents] ackcall=no musiconhold = default updatecdr=yes agent = 6301,,Agent 1 agent = 6302,,Agent 2 agent = 6303,,Agent 3 agent = 6304,,Agent 4 agent = 6305,,Agent 5 Hope this helps Regards Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queues and Transfers
Guys.. Why is it that when a call comes to a call queue and in term gets assigned to an agent, if that agent tries to xfer the call using # or any other feature, it doesn't do anything? I just hear the pleeps on the phone but asterisk doesn't intervene with the Transfer prompt. Am I missing something? Thx! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queues and Transfers
On Tue, 2005-03-15 at 03:53 -0600, Anton Krall wrote: Guys.. Why is it that when a call comes to a call queue and in term gets assigned to an agent, if that agent tries to xfer the call using # or any other feature, it doesn't do anything? I just hear the pleeps on the phone but asterisk doesn't intervene with the Transfer prompt. Am I missing something? Yes, all the information that would allow me to actually help you... How about the output from the * CLI before/during/after this attempted transfer. How about a copy of the extensions.conf (relevant portions)? That's just for starters... PS, oh okay, I'll give you a fish show application queue Regards, Adam -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queues and Transfers
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi How do you queue the incoming call ? Do you queue the call with the t option (allow the called user to transfer the calling user) ? Regards Joo Amaro Anton Krall wrote: | Guys.. Why is it that when a call comes to a call queue and in term | gets assigned to an agent, if that agent tries to xfer the call | using # or any other feature, it doesn't do anything? I just hear | the "pleeps" on the phone but asterisk doesn't intervene with the | "Transfer" prompt. | | Am I missing something? | | Thx! | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCNrUHJUm/Bor63CERArJ7AJ9abH3agaqqq12Gc4HIl+Y5wlVY/wCeM/PO 7WSe4JfZVshJVbAqPC/4r40= =EajH -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Queues and Transfers
We had the same problems with transferring calls in queues. Sometimes, after pressing the # Key twice !!, we hear Allison say Transferring. Which Phones do you use? What shows up in the cli debug? Are you using t and T options in the dial command? Regards, Guido Hecken Guys.. Why is it that when a call comes to a call queue and in term gets assigned to an agent, if that agent tries to xfer the call using # or any other feature, it doesn't do anything? I just hear the pleeps on the phone but asterisk doesn't intervene with the Transfer prompt. Am I missing something? Thx! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Queues and Transfers
Thx Adam, Ill try with |t and see what happens. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Martes, 15 de Marzo de 2005 04:08 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Queues and Transfers On Tue, 2005-03-15 at 03:53 -0600, Anton Krall wrote: Guys.. Why is it that when a call comes to a call queue and in term gets assigned to an agent, if that agent tries to xfer the call using # or any other feature, it doesn't do anything? I just hear the pleeps on the phone but asterisk doesn't intervene with the Transfer prompt. Am I missing something? Yes, all the information that would allow me to actually help you... How about the output from the * CLI before/during/after this attempted transfer. How about a copy of the extensions.conf (relevant portions)? That's just for starters... PS, oh okay, I'll give you a fish show application queue Regards, Adam -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Queues and Transfers
Jolio, no, I checked the wiki and didnt see that parameter there, but I just checked show application queue and made the necessary modifications. Thx Guys! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of João AmaroSent: Martes, 15 de Marzo de 2005 04:12 a.m.To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Call Queues and Transfers -BEGIN PGP SIGNED MESSAGE-Hash: SHA1HiHow do you queue the incoming call ?Do you queue the call with the t option (allow the called user totransfer the calling user) ?RegardsJoão AmaroAnton Krall wrote:| Guys.. Why is it that when a call comes to a call queue and in term| gets assigned to an agent, if that agent tries to xfer the call| using # or any other feature, it doesn't do anything? I just hear| the "pleeps" on the phone but asterisk doesn't intervene with the| "Transfer" prompt.|| Am I missing something?|| Thx!||-BEGIN PGP SIGNATURE-Version: GnuPG v1.2.4 (GNU/Linux)iD8DBQFCNrUHJUm/Bor63CERArJ7AJ9abH3agaqqq12Gc4HIl+Y5wlVY/wCeM/PO7WSe4JfZVshJVbAqPC/4r40==EajH-END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Queues and Transfers
On dial command yes, wtWT just in case, and it works when Im the one that originated the call, but for example, I have the same problem that you have when the call comes in thru a Zap channel. I cant make transfer to work eventhough the dial command that sent the incoming Zap call to me has wtWT. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guido Hecken Sent: Martes, 15 de Marzo de 2005 04:22 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Call Queues and Transfers We had the same problems with transferring calls in queues. Sometimes, after pressing the # Key twice !!, we hear Allison say Transferring. Which Phones do you use? What shows up in the cli debug? Are you using t and T options in the dial command? Regards, Guido Hecken Guys.. Why is it that when a call comes to a call queue and in term gets assigned to an agent, if that agent tries to xfer the call using # or any other feature, it doesn't do anything? I just hear the pleeps on the phone but asterisk doesn't intervene with the Transfer prompt. Am I missing something? Thx! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queues, CallerID, SIP and AutoDial
Hello, Currentmoment, I'vesuccessfullyput the incoming calls into Queues anddial to anidle agents.When the agents answer the calls,the agents can hear the pre-recorded message to incidate what's the service that the call is calling. But there one problem that I'm not able to make it having the Caller ID display on the X-Lite. Even I try to make a call direct transfer from Asterisk to SIP X-Lite, it's not displaying the CallerID too! Is there's any method I can show the CallerID on the X-Lite? By the way, I also want to know how can I make the Auto Dial depends on current idle agents? I know that when I put the .call files into /var/spool/asterisk/outgoing, the Asterisk will make outbound call regardness how many files there...any suggestion? Thanks in advance! Regards, R Wong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call queues
Hello I am new to asterisk I want to setup the call queues where it will ring multiple devices at the same time and send the call to the first one that is picked up. There doesn't need to be an agent login for this I don't think I just want setup so no login is required. Please help -Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call queues
On Fri, 23 Jul 2004 02:26:26 -0400, Jeremy Kenney [EMAIL PROTECTED] wrote: Hello I am new to asterisk I want to setup the call queues where it will ring multiple devices at the same time and send the call to the first one that is picked up. There doesn't need to be an agent login for this I don't think I just want setup so no login is required. Please help Check out the dial command Show application dial dial(device1device2device3) Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call queues
Quoting Jeremy Kenney [EMAIL PROTECTED]: Hello I am new to asterisk I want to setup the call queues where it will ring multiple devices at the same time and send the call to the first one that is picked up. There doesn't need to be an agent login for this I don't think I just want setup so no login is required. Please help There are several ways to accomplish this. Like the two others posts suggest - you can simply use the Dial() application directly. This will leave you with exactly the functionality you are asking for. What is does not give you is a real queue where members can join / part as they see fit (app. AddQueueMember / RemoveQueueMember). If you want to have your agents logged in from the start, you can simply define these in etc/queues.conf like SIP/phone1 or IAX2/phone1. The last option will even let you define a penalty (in etc/queues.conf). What this lacks is a persistant penalty. I've been using a little time investigating this - and I came to the conclusion that if I want persistant penalties for dynamically added members I would have to write my own wrapper in AGI. While I'm pretty much done with that part - it's not exactly a beautiful hack - but I might publish it if wanted. I will be posting on the asterisk-dev list soon - in order to get second oppinions on this implementation. Several things needs coverage - but all this in due time :) I hope you can use this - and feel free to ask into any of the above... Regards - avizion on irc.freenode.org #asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call queues
Avizion, you're joking right? -= Info about application 'AddQueueMember' =- [Synopsis]: Dynamically adds queue members [Description]: AddQueueMember(queuename[|interface[|penalty]]): The AddQueueMember function does indeed allow you to set the penalty. Too bad penalties don't work though (or maybe they work too well?) SIP/100, penalty 1 SIP/200, penalty 2 Call comes in, SIP/100 picks up Call comes in, SIP/100 is busy, but SIP/200 NEVER rings... *sigh* -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of avizion Sent: Friday, July 23, 2004 5:54 AM To: [EMAIL PROTECTED]; Jeremy Kenney Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call queues Quoting Jeremy Kenney [EMAIL PROTECTED]: Hello I am new to asterisk I want to setup the call queues where it will ring multiple devices at the same time and send the call to the first one that is picked up. There doesn't need to be an agent login for this I don't think I just want setup so no login is required. Please help There are several ways to accomplish this. Like the two others posts suggest - you can simply use the Dial() application directly. This will leave you with exactly the functionality you are asking for. What is does not give you is a real queue where members can join / part as they see fit (app. AddQueueMember / RemoveQueueMember). If you want to have your agents logged in from the start, you can simply define these in etc/queues.conf like SIP/phone1 or IAX2/phone1. The last option will even let you define a penalty (in etc/queues.conf). What this lacks is a persistant penalty. I've been using a little time investigating this - and I came to the conclusion that if I want persistant penalties for dynamically added members I would have to write my own wrapper in AGI. While I'm pretty much done with that part - it's not exactly a beautiful hack - but I might publish it if wanted. I will be posting on the asterisk-dev list soon - in order to get second oppinions on this implementation. Several things needs coverage - but all this in due time :) I hope you can use this - and feel free to ask into any of the above... Regards - avizion on irc.freenode.org #asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call queues
hehe - well - I think you missed the word persistant :) After a few hours of digging in old docs - I also found the new 3rd parameter in a more recent doc. Not sure how I missed in the first place - but I did. Never the less it did not solve my problem of having members added to my queues without using AgentLogin but still have a penalty. While goofing around I found out that adding an actual client to the queues.conf would give me a startup penalty (i.e. member = IAX2/513,50). However - this penalty is lost when the use call app RemoveQueueMember and never set again when they call AddQueueMember. So I basically just wrote a small perl script to tape through queues.conf looking up the client and add the penalty to the AddQueueMember call from extensions.conf. What I would like to have though - was the app_queue to do this for me. The code is almost there I'm sure. It just needs to lookup the penalty for a dynamically added user - if defined in queues.conf. PS: I didn't even make my wrapper look for Agent/Ext or IAX2/Ext, just Ext - but is works fine for my use atm. Only thing missing is multiple queue support - but that's easy. Are you still laughing? Please tell me better ways... I seached high and low for this - and didn't find anything. And seeing time is an issue here - I didn't want to hack the actual app_queue.c yet. But as I mentioned - I will most likely take this to -dev list and bugs.digium.com when ready. Thank you for your time and comment :) -- avizion on irc.freenode.org #asterisk Quoting Troy Settle [EMAIL PROTECTED]: Avizion, you're joking right? AddQueueMember(queuename[|interface[|penalty]]): The AddQueueMember function does indeed allow you to set the penalty. Too bad penalties don't work though (or maybe they work too well?) SIP/100, penalty 1 SIP/200, penalty 2 Call comes in, SIP/100 picks up Call comes in, SIP/100 is busy, but SIP/200 NEVER rings... Quoting Jeremy Kenney [EMAIL PROTECTED]: Hello I am new to asterisk I want to setup the call queues where it will ring multiple devices at the same time and send the call to the first one that is picked up. There doesn't need to be an agent login for this I don't think I just want setup so no login is required. Please help There are several ways to accomplish this. Like the two others posts suggest - you can simply use the Dial() application directly. This will leave you with exactly the functionality you are asking for. What is does not give you is a real queue where members can join / part as they see fit (app. AddQueueMember / RemoveQueueMember). If you want to have your agents logged in from the start, you can simply define these in etc/queues.conf like SIP/phone1 or IAX2/phone1. The last option will even let you define a penalty (in etc/queues.conf). What this lacks is a persistant penalty. I've been using a little time investigating this - and I came to the conclusion that if I want persistant penalties for dynamically added members I would have to write my own wrapper in AGI. While I'm pretty much done with that part - it's not exactly a beautiful hack - but I might publish it if wanted. I will be posting on the asterisk-dev list soon - in order to get second oppinions on this implementation. Several things needs coverage - but all this in due time :) I hope you can use this - and feel free to ask into any of the above...-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queues help
I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = queue-youarenext (You are now first in line.) queue-thereare = queue-thereare (There are) queue-callswaiting = queue-callswaiting (calls waiting.) queue-holdtime = queue-holdtime (The current est. holdtime is) queue-minutes = queue-minutes (minutes.) queue-thankyou = queue-thankyou (Thank you for your patience.) but i get: Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File queue-thereare (There are) does not exist in any format Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable to open queue-thereare (There are) (format GSM): No such file or directory -- Playing 'digits/2' (language 'en') Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File queue-callswaiting (calls waiting.) does not exist in any format Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable to open queue-callswaiting (calls waiting.) (format GSM): No such file or directory -- Hold time for queue1 is 0 minutes -- Told Local/[EMAIL PROTECTED],2 in queue1 their queue position (which was 2) Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File queue-thankyou (Thank you for your patience.) does not exist in any format Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable to open queue-thankyou (Thank you for your patience.) (format GSM): No such file or directory I've downloaded the asterisk additional sound files and installed them and tried to use those instead and I can play them with the playback command, but they dont work here either. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Queues help
queue-youarenext = queue-youarenext Like that.. remove the quotes and all the crap at the end. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Thursday, July 15, 2004 4:48 PM To: Asterisk Subject: [Asterisk-Users] Call Queues help I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = queue-youarenext (You are now first in line.) queue-thereare = queue-thereare (There are) queue-callswaiting = queue-callswaiting (calls waiting.) queue-holdtime = queue-holdtime (The current est. holdtime is) queue-minutes = queue-minutes (minutes.) queue-thankyou = queue-thankyou (Thank you for your patience.) but i get: Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File queue-thereare (There are) does not exist in any format Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable to open queue-thereare (There are) (format GSM): No such file or directory -- Playing 'digits/2' (language 'en') Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File queue-callswaiting (calls waiting.) does not exist in any format Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable to open queue-callswaiting (calls waiting.) (format GSM): No such file or directory -- Hold time for queue1 is 0 minutes -- Told Local/[EMAIL PROTECTED],2 in queue1 their queue position (which was 2) Jul 15 14:38:55 WARNING[1267933760]: file.c:464 ast_openstream: File queue-thankyou (Thank you for your patience.) does not exist in any format Jul 15 14:38:55 WARNING[1267933760]: file.c:752 ast_streamfile: Unable to open queue-thankyou (Thank you for your patience.) (format GSM): No such file or directory I've downloaded the asterisk additional sound files and installed them and tried to use those instead and I can play them with the playback command, but they dont work here either. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queues, Call groups
Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queues
On Tue, 10 Feb 2004, Jonathan Stanton @ Home wrote: Any ideas / sugestions welcome. Having the queue calls delivered to an agent login would appear to be the easiest way to do it - just log in the agent on any phone you like, and calls will be diverted to that phone. As another poster noted, one little gotcha is that if your announcement file is missing, the calls will be disconnected when you try to answer them. -- Jon Stockill [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Queues
Title: Message
Here
is some config that I cooked up. It may be a little rough around the
edges, and it incorporates multiple users.
exten
= *801,1,Answerexten =
*801,2,SetVar(temp=${loggedin${CALLERIDNUM}})exten =
*801,3,GotoIf($[${temp} = 1]?50:)exten = *801,4,GotoIf($[${CALLERIDNUM}
= 11]?11)exten = *801,5,GotoIf($[${CALLERIDNUM} = 12]?12)exten =
*801,6,GotoIf($[${CALLERIDNUM} = 13]?13)exten =
*801,7,GotoIf($[${CALLERIDNUM} = 15]?15)exten =
*801,8,Playback(beep)exten = *801,9,Hangupexten =
*801,11,Goto(bjqueue,1,1)exten = *801,12,Goto(tonyqueue,1,1)exten
= *801,13,Goto(wendyqueue,1,1)exten =
*801,15,Goto(danqueue,1,1)
exten
= *801,50,GotoIf($[${CALLERIDNUM} = 11]?61)exten =
*801,51,GotoIf($[${CALLERIDNUM} = 12]?62)exten =
*801,52,GotoIf($[${CALLERIDNUM} = 13]?63)exten =
*801,53,GotoIf($[${CALLERIDNUM} = 15]?65)exten =
*801,54,Playback(beep)exten = *801,55,Hangupexten =
*801,61,Goto(bjqueue,2,1)exten = *801,62,Goto(tonyqueue,2,1)exten
= *801,63,Goto(wendyqueue,2,1)exten =
*801,65,Goto(danqueue,2,1)
[bjqueue]exten = 1,1,Answerexten = 1,2,Wait(1)exten
= 1,3,Playback(agent-loginok)exten =
1,4,SetGlobalVar(loggedin11=1)exten =
1,5,AddQueueMember(tech,SIP/111)exten = 1,6,Hangup
exten
= 2,1,Answerexten = 2,2,Wait(1)exten =
2,3,Playback(agent-loggedoff)exten =
2,4,SetGlobalVar(loggedin11=0)exten =
2,5,RemoveQueueMember(tech,SIP/111)exten =
2,6,Hangup
The
other users are setup pretty much the same as bjqueue. I hope this
helps.
B.
J.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Stanton @ HomeSent: Monday, February 09, 2004 18:28To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Call
Queues
Dear all,
I am one of the people who answer the FWD welcome
line. Since I don't want my phone ringing at 2am I have the 5 number
routed to a call queue. Currently I have 2 extentions 271 which will log
my phone into the queue and 270 to log it out. What I want to know is...
Is there a way to do this with just one exten? I have tried with the
following :
exten =
270,1,AddQueueMember(pulverWelcome)exten = 270,2,AGI(cepstral.agi,You
have just joined the Welcome queue)exten = 270,3,Hangupexten =
270,102,RemoveQueueMember(pulverWelcome)exten =
270,103,AGI(cepstral.agi,You have just left the Welcome queue)exten =
270,104,Hangup
It works when I log in but when I try to log out
it crashes (403 error). I know that the AddQueueMember returns -1, so
why does it not jump to priority 102 instead?
Any ideas / sugestions
welcome.
Jonathan
[Asterisk-Users] Call Queues
Dear all, I am one of the people who answer the FWD welcome line. Since I don't want my phone ringing at 2am I have the 5 number routed to a call queue. Currently I have 2 extentions 271 which will log my phone into the queue and 270 to log it out. What I want to know is... Is there a way to do this with just one exten? I have tried with the following : exten = 270,1,AddQueueMember(pulverWelcome)exten = 270,2,AGI(cepstral.agi,You have just joined the Welcome queue)exten = 270,3,Hangupexten = 270,102,RemoveQueueMember(pulverWelcome)exten = 270,103,AGI(cepstral.agi,You have just left the Welcome queue)exten = 270,104,Hangup It works when I log in but when I try to log out it crashes (403 error). I know that the AddQueueMember returns -1, so why does it not jump to priority 102 instead? Any ideas / sugestions welcome. Jonathan smime.p7s Description: S/MIME cryptographic signature
[Asterisk-Users] Call Queues
I have setup AgentCallbackLogin and the agents have been logged in successfully. However when calls are queued and an agent picks up the call. It just hang up the call. On the command console it does say the agent agent 1001 hang up on customers. they must be pissed off. I agreed. My queues.conf file: [agents] ackcall=no agent = 1001,1001,xx ss My queues.conf file: [incoming] announce = incoming strategy=ringall musice = default member = Agent/1001 member = Agent/1002 My extensions.conf : exten = 28,1,AgentCallbackLogin(|@local) exten = 29,1,Queue(incoming) In order to annonce to agent the correct queue does it have to have a gsm file to playback the name of the queue ie incoming in this case? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Call Queues
I have setup AgentCallbackLogin and the agents have been logged in successfully. However when calls are queued and an agent picks up the call. It just hang up the call. On the command console it does say the agent agent 1001 hang up on customers. they must be pissed off. I agreed. My queues.conf file: [agents] ackcall=no agent = 1001,1001,xx ss My queues.conf file: [incoming] announce = incoming strategy=ringall musice = default member = Agent/1001 member = Agent/1002 My extensions.conf : exten = 28,1,AgentCallbackLogin(|@local) exten = 29,1,Queue(incoming) In order to annonce to agent the correct queue does it have to have a gsm file to playback the name of the queue ie incoming in this case? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Call queues for phone operator
Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member is my op. into technology SIP/operator So my queues.conf should be : [general] [default] [operatorq] music = default announce = from-queue ;context = qoutcon timeout = 20 retry = 5 ;maxlen = 0 member = SIP/operator and when a call arrives, dial the operator and if he's busy, fire up app_queue . So what I expect, when the operator hangs up, his phone will automagically rings playing the announce from-queue and bridge it with the call that's waiting. So, I'm correct? Anyone experienced that or could give me a better way to handle that? Thanks a lot, matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call queues for phone operator
But he didn't think about agent. Just a regular SIP phone. It should be in general like the original author of this thread thinks. Besides it's easy to test so why not to test it :) Martin On Fri, 13 Jun 2003, TC wrote: Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. Thats correct So I want to ask if I'm right or wrong: and when a call arrives, dial the operator and if he's busy, fire up app_queue . NO agents log into an agent q their phone is OFF-HOOK always thus if you Dial that agent ext it is always busy So what I expect, when the operator hangs up, his phone will automagically rings playing the announce from-queue and bridge it with the call that's waiting. the agent will just hear beep beep the optional announcement on the handset/speakerphone or headset then the inbound caller is bridged ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call queues for phone operator
http://www.digium.com/asterisk_handbook/agentlogin_queues.html Brancaleoni Matteo wrote: Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member is my op. into technology SIP/operator So my queues.conf should be : [general] [default] [operatorq] music = default announce = from-queue ;context = qoutcon timeout = 20 retry = 5 ;maxlen = 0 member = SIP/operator and when a call arrives, dial the operator and if he's busy, fire up app_queue . So what I expect, when the operator hangs up, his phone will automagically rings playing the announce from-queue and bridge it with the call that's waiting. So, I'm correct? Anyone experienced that or could give me a better way to handle that? Thanks a lot, matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
