Re: [asterisk-users] Best way to update ever changing dialplans
Depends on how complex the generated dial-plan is.If it is in a monitored production environment, you could have an ARI app and just let it handle it all, especially if it needs to make complex decisions that end in simple choices (like: detecting the caller, asking them some questions but finally routing to queue A versus B). 2018-06-25 18:54 GMT+02:00 Dovid Bender : > I am working on a system where I connect to an external API and based on > what it gives me I generate the Asterisk dial plan accordingly. I am > thinking about my different options and wanted feedback from others on how > to best do it. > 1) Generate conf files for Asterisk - This seems the easiest but then I will > be doing a dial plan reload on all of my dial plan for handful of lines of > code. The plus side is once reload is don the dial plan is in memory. > 2) Using real time + mysql - Seems like an overkill to have mysql running > taking resources for a few lines. > 3) Using real time + sqlite3 - This seems like the best option but then we > go to disk every time there is a call. > > Any other options that I am not thinking of? > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A survey on Asterisk-based call-centres - Help needed
Hi all, I am running a survey of Asterisk-based call-centres, to understand what they are doing now and how they expect to grow in the future. Results will be presented by yours truly in October at the Astricon in Orlando, but you can also sign up to receive them when they will be ready. See http://www.digium.com/blog/2018/06/27/asterisk-contact-center-survey-results-will-be-interesting/ So, if you run a call-center based on Asterisk, or you have customers doing it, why not letting the community know what you are doing and what you wish for? it only takes 5 minutes, but it’s a way for your voice be heard. You can find the survey at: https://www.queuemetrics.com/callcenter-survey-2018.jsp?lid=B083 Best, lenz -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tracking Music-on-Hold on call queues
Hi all, we have a little tool that tracks Music-on-Hold events for call queues by listening to AMI events. This is quite useful for reporting so, as the tool is free to use and does not depend on our QueueMetrics Call Center suite, I thought I'd announce it in here as well. If anyone is interested, you can find a post here: https://www.queuemetrics.com/blog/2017/03/22/TrackingMOH/?lid=A002 Comments welcome :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com WombatDialer - next generation predictive dialer - http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pet project: one step Asterisk compile on Centos 7
2016-06-14 17:44 GMT+02:00 Tzafrir Cohen: > > 1. Asterisk basically has such a script inside. It is - as you say - inside. This is outside and does the download for you. > 2. Asterisk has an RPM package. An RPM package is exactly a reproducible > build (listing dependecies, and such). It's true. They are very interesting, especially if you are a historian of software. http://packages.asterisk.org/centos/ If you need something less, say, "vintage", you may need to compile it yourself. > 3. You are reinventing RPM. Badly. Do you people really want to run: >- As root >- A huge blob nobody can inspect >- that is executable, and hence has tons of places to add nice hooks > in? > > Learn how to use rpmbuild. I personally happen to have shipped RPMs for about 10 years now. But building a RPM might be overkill if you are deploying a test, throwaway box, or just once for a Docker image. Of course I would not use this as an RPM substitute, and if I were to use something like this I'd fork it or at least read it (it is maybe 20 lines). YMMV. And IIRC there is more places to ship "nice hooks" into a binary you ship as an RPM than in a shell script that does what you would manually from the terminal! l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
[asterisk-users] Pet project: one step Asterisk compile on Centos 7
Hi all, I thought I'd share I script I made (based on some of Leif's works) that lets you download, compile and install Asterisk all in one go; and then removed the dev tools used. We use it quite a bit to provision systems using Ansible, but it is easier than remembering everything every time even if you are using a shell. At the moment I have scripts for Centos 7 and Asterisk 13, but plan to port them to other versions of Asterisk as there is a need to do so. Contribs welcome! Project located at https://github.com/l3nz/CompileAsteriskPBX Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] [SOLVED] AMI issue with Filter
D'oh moment - filters do work, but you can only set one at a time. By reading the docs, I mistakenly assumed that you could set many of them at once. Thanks l. 2016-05-19 12:54 GMT+02:00 Lenz Emilitri <lenz.lo...@gmail.com>: > Hello all, > I am trying to use the Filter action in AMI to make AMI less chatty by > blacklisting some events; and I must be doing something wrong, because > if I send something like: > > Action: Filter > ActionID: AID563116752-152218 > Operation: Add > Filter: !Event: VarSet* > Filter: !Event: ExtensionStatus* > Filter: !Event: NewAccountCode* > Filter: !Event: NewCallerid* > Filter: !Event: Newexten* > Filter: !Event: RTCPSent* > > I still get plenty of: > > Uniqueid: 1463577738.539 > Extension: 201 > Channel: SIP/200-00a1 > Context: from-internal > Event: Newexten > Application: Set > Privilege: dialplan,all > AppData: ADMINCODE=15 > Priority: 1 > > I get them with or without the trailing *. > > I am testing this on Asterisk 11 and 13, so I must be doing something > wrong - but what? :-) > Thanks > l. > > > > > -- > Loway - home of QueueMetrics - http://queuemetrics.com > Try the WombatDialer auto-dialer @ http://wombatdialer.com -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Call File - CPU spikes
If you are on 13 it would likely be easier to use ARI directly? l. 2016-05-11 22:52 GMT+02:00 Bryant Zimmerman: > I am working on a project that we are seeing a 100% CPU spike when we move > 50 calls files to the folder. > > We are running pjsip and asterisk 13..It holds the spike for several minutes > Are there any tunable that may help with this? > > > Thanks > Bryant > > -- > _ > -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
[asterisk-users] AMI issue with Filter
Hello all, I am trying to use the Filter action in AMI to make AMI less chatty by blacklisting some events; and I must be doing something wrong, because if I send something like: Action: Filter ActionID: AID563116752-152218 Operation: Add Filter: !Event: VarSet* Filter: !Event: ExtensionStatus* Filter: !Event: NewAccountCode* Filter: !Event: NewCallerid* Filter: !Event: Newexten* Filter: !Event: RTCPSent* I still get plenty of: Uniqueid: 1463577738.539 Extension: 201 Channel: SIP/200-00a1 Context: from-internal Event: Newexten Application: Set Privilege: dialplan,all AppData: ADMINCODE=15 Priority: 1 I get them with or without the trailing *. I am testing this on Asterisk 11 and 13, so I must be doing something wrong - but what? :-) Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
[asterisk-users] Sample Docker images for Asterisk available
Hello all, I created a set of Docker images running Asterisk and exposing AMI / ARI ports that i found to be quite useful for ARI / AMI development and regression. As they are based on Docker with whaleware, adding new configuration files to roll your own dialplan / queues / voicemail etc is pretty easy. And you can run quite a lot on the same box to simulate clusters. There is no SIP / RTP configured at the moment. See https://github.com/l3nz/whaleware/blob/master/examples/asterisk-load-test/README.md Maybe somebody else might find them useful. There is Asterisk 1.8, 11, 12 and 13. Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Sample Docker images for Asterisk available
If you need to add files, either you create a new Docker image that inherits from one of the images and add properties files to /ww/files, or just copy the files into a docker instance and do a docker exec to have Asterisk reload them. See e.g. http://stackoverflow.com/questions/22907231/copying-files-from-host-to-docker-container Make sure you have a look at what whaleware does, as it acts as a good template and manages a number of things for you (eg configuration). Just .02/chf l. 2015-04-23 8:47 GMT+02:00 Guenther Boelter gboel...@gmail.com: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 04/23/2015 02:03 PM, Lenz Emilitri wrote: Hello all, I created a set of Docker images running Asterisk and exposing AMI / ARI ports that i found to be quite useful for ARI / AMI development and regression. As they are based on Docker with whaleware, adding new configuration files to roll your own dialplan / queues / voicemail etc is pretty easy. And you can run quite a lot on the same box to simulate clusters. There is no SIP / RTP configured at the moment. See https://github.com/l3nz/whaleware/blob/master/examples/asterisk-load-t est/README.md Maybe somebody else might find them useful. There is Asterisk 1.8, 11, 12 and 13. Thanks Great, will try it out tonight ... Thanks - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -BEGIN PGP SIGNATURE- Version: GnuPG v2 iQIcBAEBAgAGBQJVOJVpAAoJENexF5oIz3BC08oQAJMOx64PLaVQVtJYDyZH78Sl V134Rgv/Fq/1Udm4dZlt4Dooo3cEYPCd+WBC70VU+4hNegSgv6xZxGSMXyymPT/R SrjL/4jMJ/9S8uanIRdiZbAoKRByDacw/CgzF1CO4Jd9DQTu1L/Smz3VC5DhU0Cg ZOVdJW49FRv8TrfYiLp6YzdKgf16/qwhpAif7RGNoms4Ixfg3T8r72sKHisl+0v4 7qQ4a6t3nByo54WIH3IvAHrmvYTSiL4DtQS2bTddhZshw5isWOQ1x6gfwE+ecefC O9zmsyUqvKFBg+zaCuo6lShwpr7t2wFNQKtH6ndVbikAR1oUdIJ7pMPLbgVTVCJ7 TrOKvkllvs8b5l/k/ahjRQDHgZ6eZwnatnp0Woao8rH6u7VG1yWq4g8d4wcxcMp7 exK1ex3yLs3ZWuZuvif9ivUj60+RWqiSkJZCvMSqNO0hMpdzAYZuT5F1NfFNt7tU OvOH15jPvHr7jrgMo+XljIxBgg1RUHuM4NZzUg3N/0OTMthWhC5mhn8YODfctrRq QFPYACEgv969N/3jA2JoIhOkvyLDRoovjem3RNsrHiGKzSyvyjlFm7q0sLz5LRTL DR+bd95eqg/+78rQLxpSdGgLgbfN7GK1mSaDhLUY6bmlTSdfxQI4nF3xtz3bLqYm 7d5SrxB/ct0RmjgV88GP =FSuK -END PGP SIGNATURE- -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
[asterisk-users] Tutorial: compiling and installing Asterisk 13
Hi all, I just prepared a little tutorial on installing Asterisk 13 on CentOS 6.5 64-bit. See http://astrecipes.net/index.php?n=668 Hope you like. :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] pull a call from a queue
What you usually do is to transfer the call to a second VIP queue. This can be done in the free version of our QM or I'm sure there are other products as well :) 2014-06-13 20:15 GMT+02:00 Adam Moffett adamli...@plexicomm.net: We have a queue monitoring application running so we can see the caller ID of callers in a queue. If we see a VIP in the queue, is there any method to force that call to be first in line? If there's a softphone, or queue managing application already written that does this, I'd love to know. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Open Source Asterisk Polling Solution
Our Wombat is not open source but is free to use for small systems and will be very trivial to set up for such a task. And if you ever need to grow up, you're covered. :) l. 2014-04-21 19:45 GMT+02:00 Nick Cameo sym...@gmail.com: Hello Everyone, We are looking for a simple open source auto dialer with polling capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free version etc Something that can support 10 channels, and is stable would be greatly appreciated. If this can be simply implemented using asterisk and call folder, even better PS Our preferred version of * is 1.8.x Kind Regards, Nick from Toronto. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Numbers hackers call
http://en.wikipedia.org/wiki/Telephone_numbers_in_Israel Looks like it a mobile in Palestine - sure someone from Israel can tell us more 2014-03-26 16:05 GMT+01:00 Michelle Dupuis mdup...@ocg.ca: I see a lot of attempts by hackers to call 00972595301123 or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
[asterisk-users] Couple of new tutorials on asterisk 12 and ARI
Hi all, I put together a couple of new tutorials on compiling Asterisk 12 with PJSIP on CentOS 6.5 and test-driving ARI on the same box. You can find them at: http://astrecipes.net/index.php?q=AstRecipes/Compiling%20Asterisk%2012%20on%20CentOS%206.5 and http://astrecipes.net/index.php?q=AstRecipes/Getting%20started%20with%20ARI Comments welcome and happy holidays! :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Call Queue advise
Most likely the feature can be obtained with the wrap-up time or pausing the agent. I agree on removing ring-all if possible (Though a number of clients want it in smaller set-ups, and I know there is nothing you can do to make them change their mind). 2013/12/11 Paul Belanger paul.belan...@polybeacon.com: On 13-12-09 06:47 PM, Bryan Anderson wrote: I have a call queue that rings about 15 users and they are wanting to set it up so that the last person to answer a call doesn't ring on the next incoming call. What would be the best way to handle this? I have been looking at the strategies and none of those seem to be right for this. My current thoughts are probably a macro that places a penalty on the user tell the next call is answered. Any advice for this would be greatly appreciated. You have agents that log into a queue that don't want to get calls? Is that what you are saying? Options 1 - log the agent out, they don't get the next call. Option 2 - Set up weights for your agents, as answer a new call, increment then up so they don't get the next. Either way, I see issues with the setup. Best ways is to rethink your queue strategy and stop using ring all. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Calls Recording Solution
We have a number of clients using OrecX and they are quite happy about it. l. 2013/10/22 bilal ghayyad bilmar...@yahoo.com: Hello; I am looking for calls recording solution to do recording based on the network traffic .. The solution to be competitive and appreciate if it is open source .. Any suggested one? Regards Bilal -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
[asterisk-users] QueueWiz - a free call-center simulator tool for Asterisk
Hello all, next week it's Astricon 10 time, so we thought we'd create something that the community could like and use for free. It's a pretty effective tool if you run a call-center or plan to run one. QueueWiz is the first free web app for interactive, quick and accurate call center sizing, cost and revenue simulation. Insert your data with the intuitive interface, measure traffic intensity, expected wait times, agents' engagement, revenue per call and per agent and even hourly margins. Save your simulation and share it via email or social media. Completely free of charge - no string attached - try it at http://queuewiz.queuemetrics.com Have a great day and see you next week. l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Queue Management
This should happen automatically - not sure what you want to do. l. 2013/9/26 akhilesh chand omakhileshch...@gmail.com: Dear All, I have six different campaign and 5 different agent have login on that campaign.Same thing i have done using agi and database,i never use queue management on this scenario. Agent can also shuffling one campaign to anther campaign. Now i want to do some work with queue.I want to use single queue to managing this. Eg: campaign Agent Login A a_1,a_3 (In campaign A 2 agents are login) B a_2,a_1 (In campaign B 2 agents are login) C a_3,a_1,a_4 (In campaign C 3 agents are login) D a_4,a_5,a_3 (In campaign D 3 agents are login) E a_1,a_3,1_2 (In campaign E 3 agents are login) Fa_5,a_4 (In campaign F 2 agents are login) When a call come to campaign A that call goes to agent a_1 or a_3 not goes to other campaigns agents. Regards Akhilesh -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
[asterisk-users] Astricon - let's talk call centers?
Hi list, I know it's a bit OT, but for those who will be at the Astricon, we are organizing a very informal meeting (maybe in front of a pint or two) to talk about Asterisk for call-centers. No marketing or anything - just a way to exchange ideas and meet f2f. I created a facebook group to organize it - see https://www.facebook.com/groups/507826572618269/ See you in Atlanta! l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Pull call out of queue
You could transfer to a dead-end extension that plays MOH and then transfer it back somewhere else. Or to a queue with no agents on (if you are using queues, most likely you are already monitoring queues, so this may make your workflow easier to live with). l. 2013/9/6 Todd R. tjrl...@live.com: Trying to figure out the best way to pull an active call out of a queue by unique id and put it on hold. I don't want to put it on hold on the agent's phone but I want it to be pulled away from the agent's phone and into Asterisk limbo somewhere. Shortly after I want to pull the same call out of limbo and redirect it back to either the same agent or another. I was thinking about call parking but, I think parking is more than I need and it potentially introduces more complications. I will be doing this through the manager interface on Asterisk 1.8.x. Any ideas, thoughts or help would be greatly appreciated. Thanks in advance for any help. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
[asterisk-users] Setting different caller-id for second leg of the Originate
Hello all, I would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: 123 123 Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1@mycontext) to something different. How do I do that? Would it be enough to change the caller-id as soon as the call is successfully connected? Thanks for any pointers, l. PS: I Know one can easily do this by editing the dialplan at 1@mycontext, but this is something we cannot do now. -- _ -- 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] Need input on scalable system design...
Hi Greg, I am aware of a couple of solutions that come prepackaged and offer distributed queues for Asterisk. One of them, that seems to work well and reliably, is the one from Raynet. I am sure there are more. On the other side, I have seen a number of in-house solutions where you basically have a daemon polling queues statuses and redirecting calls based on the relative wait times. Rough but effective, and can be deployed easily. About recordings, my suggestion would be to use something to offload them right from the servers, like Oreka. Have a number of large clients using it and they are quite happy (plus, the guys supporting it are superb). Just my two cents, l. 2013/8/27 Gregory Malsack gmals...@coastalacq.com Hey All, Growing call center. Currently at about 200 call center staff, running about 1000 calls per hour. Gearing up to double that. Not too sure that a single server will support that growth. So, I'm trying to come up with ways to scale the system and still maintain a simplistic design. So I'd like to bounce some ideas around. Currently I am running on a Dell 1950, dual quad core 2.33ghz xeons, with 16gb ram, and 2 tce400p cards. This server is managing the full load of the company. We are recording all calls, running ivr, queues, cdr, cel, and web for reporting. I currently have another 1950 of the exact same specifications as a cold spare. Here's where you can see drawings of my current connectivity and an optional connectivity I'm contemplating... http://www.paydaysupportcenter.com/current.pdfhttp://www.linkedin.com/redirect?url=http%3A%2F%2Fwww%2Epaydaysupportcenter%2Ecom%2Fcurrent%2Epdfurlhash=qLsB_t=tracking_anet http://www.paydaysupportcenter.com/option.pdfhttp://www.linkedin.com/redirect?url=http%3A%2F%2Fwww%2Epaydaysupportcenter%2Ecom%2Foption%2Epdfurlhash=CJG1_t=tracking_anet As you can see I currently have a separate sql server and a separate storage server for the call recordings. This is all working fine. However, I'm thinking for scalability I should be looking to migrate to a configuration similar to the one in option.pdf. Where I have a VOIP gateway server that simply relays traffic and possibly can do some load balancing or intellegent routing. But nothing more then that, and possibly a second one of these online as a hot failover. Then have separate sql, storage, (i forgot it in the pic) web, and asterisk servers behind that on separate dedicated network. Here's my dilemma though, how do I balance the load across multiple machines for scalability... Since 95% of our calls come into queues, I need to be able to maintain queue stats and presence across all of the servers. Thus far, I've got everything except the extensions.conf file into the mysql database. I thought about setting up 2 servers, 1 for sales, and 1 for customer service, then possibly break out each call queue to it's own server as things grow. Just not sure if that's the right way to go. Then regarding extensions.conf, I've read that it too can be placed in the sql database and accessed via switch. however it's resource intense, so now I'm thinking of maybe putting that file on the nfs server for all of the boxes to read from. As for the design of that file, I was kind of thinking of a modular design within the file using various goto's and gosubs. Our business model is based on affiliates and corporate marketing, so we have a ton of did's that follow the same call flow with minor modifications in some variables, as well as variations in call flow, and hours of operation. Thus the modular design of the call flow. Then the primary inbound context would simply be a list of did's pointing to a goto with a list of the variations and variables for the did. Ok, now that I've melted your brains thoughts? Thanks all in advance for the discussion... Greg -- Greg -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Kepress while on Queue
Yes it will work. One interesting option here is adding to the MOH an invitation to exit and leave your number and the CC will call you back. Helps you smooth the load during peak times, reduces staff and everyone wins :) l. 2013/8/27 Gopalakrishnan N gopalakrishnan...@gmail.com Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)
You need to do this when the call connects. If you can do this within a couple of seconds, this is usually good enough to be usable (that's what we do on the QueueMetrics agents pages). Thanks l. 2013/8/3 Timothy Smith timotsm...@gmail.com Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate
Looks correct to me 2013/6/19 Grant Bagdasarian g...@cm.nl Hello, ** ** I’d like to use the AMI interface to originate a call to a context in a dialplan, and handoff the dial control to the context. ** ** Whenever I execute the below action, the recipient does ring, but when I answer it dials the recipient again. I believe this is because once answered the system is going to execute the Context/Exten/Prio in the Originate action? ** ** Action: Originate Channel: Local/outbound1@originateDialContext CallerID: 00311234567 Context: originateDialContext Exten: outbound1 Priority: 1 Variable: recipient=0031612345678 Timeout: 1 ** ** [originateDialContext] exten = outbound1,1,Wait(1) exten = outbound1,n,Set(recipient=${recipient}) exten = outbound1,n,Dial(SIP/${recipient}@originateChannel) ** ** Anyone have an idea how to fix this? -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Queue Limit Callers
You should have different sets of agents logged in to different queues and you should have a monitor to move them from one queue to the other based on incoming traffic. l. 2013/6/17 Shanavaz E A shanava...@yahoo.com Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I have two Queues - Sales Booking I have 12 Agents who are added to both the queues Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales Queue. Only 8 calls in the Booking Queue should hit the Agents and the other 4 calls should remain in hold. 4 calls in the Sales Queue should hit the other 4 agents and the other 2 call should be in hold. Means at a time a maximum of 8 Booking calls only should hit the agents and 4 Sales Calls only should hit the agents. If number of logged in agents are less, proportionally the number of call limit should be reduced. For example, if there are only 10 agents, 7 Booking Calls should hit and 3 Sales calls should hit. The idea is that all agents should be able to answer calls in both queues in rotation. Otherwise its possible to add some agents to booking queue and other agents to sales queue. But thats not what is required. Kindly help if there is some idea to implement this. Regards Shanavaz. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?
Looks yummy! http://phono.com/webrtc 2013/5/31 Adnan 112linuxstockh...@gmail.com Voxeo/Phono webrtc. /Adnan On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri lenz.lo...@gmail.comwrote: Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
[asterisk-users] WebRTC softphone for Asterisk - any suggestion?
Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] dial and bridge
Thanks all for your help, in the end I was able to do something like: Action: Originate Channel: Local/300@from-internal/n Application: MusicOnHold Async: 1 As soon as this connects, the callee hears MOH. I get the channel out via AMI events and start another call: Action: Originate Channel: Local/301@from-internal/n Application: Bridge Data: Local/300@from-internal-aa8c;1 Async: 1 when this connects, it is immediately bridged to the first callee. I just have to keep track of errors and hang up the first call if the seconds does not go through. Thanks a lot! l. 2013/5/15 Dan Cropp d...@amtelco.com You could use AsyncAGI to achieve this. ** ** Originate the first call (passing in some unique identifier as a variable), then using AMI you will see the channel data. When you see an Event: AysncAGI for that channel (with that id, you have control of the call). Send a Dial Action telling it to dial the call and bridge them together if the person answers. If they don’t answer, you will be notified and can do something with the original call (play a message, hangup, etc). If they are bridged, you can see how long, etc. ** ** Setup an extension, naming it something like patching ** ** exten = patching,1,AGI(agi:async) ** ** Action: Originate Channel: Local/300@from-internal Async: 1 Exten: 1 Context: patching Data: 1973 Variable: YourUniquePatchID=1234 ** ** ** ** Using AsyncAGI and AMI, you can have full control of the call. You do have to setup a very simple dial plan so Asterisk knows you are using AsyncAGI to control the call. ** ** Have a great day! Dan ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Tuesday, May 14, 2013 11:16 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] dial and bridge ** ** Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a channel format, like SIp/1234 or Local/1234@ext) and park it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. ** ** As a requirement, I cannot use the dialplan as an end-point (as I cannot change it) but need to use the AMI only. ** ** I tried doing something like: ** ** Action: Originate Channel: Local/300@from-internal Async: 1 Application: Wait Data: 1973 ** ** So that the call goes to 300 and then basically stays there forever, and then I dial again: ** ** Action: Originate Channel: Local/500@from-internal Async: 1 Application: Wait Data: 1973 And then try to bridge the results, but it does not seem to work. What I would like to do would be more on the lines of: ** ** Originate call 1 and park it (using a park or waiting) Originate call 2 and bridge it immediately to call1 (using the Application part) ** ** But maybe I am missing something? is there anybody who has better suggestions? ** ** Thanks l. ** ** ** ** ** ** ** ** ** ** ** ** -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] dial and bridge
Hi Mitul, I agree that the dialplan way is easier, but it's a client requirement to avoid using it. I was wondering if there was a way to send a call directly to a parking slot right from the originate, because that is cheaper than running conferences, and then joining the second call right to the parked call, so that all we have to do is two originates. l. 2013/5/14 Mitul Limbani mi...@enterux.in Dial first call and put it into a conference, then dial second call and put him into same conference to bridge both. However dial plan way is much more simpler. Mitul On Tuesday, May 14, 2013, Lenz Emilitri wrote: Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a channel format, like SIp/1234 or Local/1234@ext) and park it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the dialplan as an end-point (as I cannot change it) but need to use the AMI only. I tried doing something like: Action: Originate Channel: Local/300@from-internal Async: 1 Application: Wait Data: 1973 So that the call goes to 300 and then basically stays there forever, and then I dial again: Action: Originate Channel: Local/500@from-internal Async: 1 Application: Wait Data: 1973 And then try to bridge the results, but it does not seem to work. What I would like to do would be more on the lines of: Originate call 1 and park it (using a park or waiting) Originate call 2 and bridge it immediately to call1 (using the Application part) But maybe I am missing something? is there anybody who has better suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] dial and bridge
Hi Warren, the problem is that all I have is two channels, so the specs might be join SIP/123 and SIP/345 not join SIP/123 to 456@from-internal. They might be Local channels, but this should be able handle the general case. The reason why I have channels and not ext@ctxt is that I read them live from the AMI itself. any idea on how to do this? Thanks l. 2013/5/14 Warren Selby wcse...@selbytech.com On Tue, May 14, 2013 at 11:16 AM, Lenz Emilitri lenz.lo...@gmail.comwrote: Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a channel format, like SIp/1234 or Local/1234@ext) and park it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. Why not just originate from one extension to the other? Something like this (not tested): Action: Originate Channel: Local/300@from-internal Context: from-internal Exten: 500 Timeout: 30 Should dial extension 500 in the from-internal context after the call to 300@from-internal is answered. Meaning, the person at 300@from-internalwould have their phone ring, they'd pick it up, and then they'd hear ringing on the line as asterisk then dialed extension 500@from-internal. -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] dial and bridge
I never actually used parking, but should it work if I call the Park application as the second leg of the Originate (w/o going through the dialplan)? I dont seem to be able to make it work. l. 2013/5/15 Mitul Limbani mi...@enterux.in The dial n bridge might work, but there ain't indefinite wait in that scenario. Direct calls to parking you might try Local(70X@from-internal) but I m not sure if this method works reliably. The method I mentioned is used by vicidial and it works flawlessly, yes it comes with some computing load, however you can try the newer ConfBridge app to see if its cheaper. Mitul On Wednesday, May 15, 2013, Lenz Emilitri wrote: Hi Mitul, I agree that the dialplan way is easier, but it's a client requirement to avoid using it. I was wondering if there was a way to send a call directly to a parking slot right from the originate, because that is cheaper than running conferences, and then joining the second call right to the parked call, so that all we have to do is two originates. l. 2013/5/14 Mitul Limbani mi...@enterux.in Dial first call and put it into a conference, then dial second call and put him into same conference to bridge both. However dial plan way is much more simpler. Mitul On Tuesday, May 14, 2013, Lenz Emilitri wrote: Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a channel format, like SIp/1234 or Local/1234@ext) and park it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the dialplan as an end-point (as I cannot change it) but need to use the AMI only. I tried doing something like: Action: Originate Channel: Local/300@from-internal Async: 1 Application: Wait Data: 1973 So that the call goes to 300 and then basically stays there forever, and then I dial again: Action: Originate Channel: Local/500@from-internal Async: 1 Application: Wait Data: 1973 And then try to bridge the results, but it does not seem to work. What I would like to do would be more on the lines of: Originate call 1 and park it (using a park or waiting) Originate call 2 and bridge it immediately to call1 (using the Application part) But maybe I am missing something? is there anybody who has better suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
[asterisk-users] dial and bridge
Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a channel format, like SIp/1234 or Local/1234@ext) and park it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the dialplan as an end-point (as I cannot change it) but need to use the AMI only. I tried doing something like: Action: Originate Channel: Local/300@from-internal Async: 1 Application: Wait Data: 1973 So that the call goes to 300 and then basically stays there forever, and then I dial again: Action: Originate Channel: Local/500@from-internal Async: 1 Application: Wait Data: 1973 And then try to bridge the results, but it does not seem to work. What I would like to do would be more on the lines of: Originate call 1 and park it (using a park or waiting) Originate call 2 and bridge it immediately to call1 (using the Application part) But maybe I am missing something? is there anybody who has better suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
[asterisk-users] amiDebugger - might make your life easier if you program through the AMI
Hi all, I have been playing with the AMI quite a bit lately - mostly debugging WombatDialer in production, but that's a different story - and I have been frustrated by the lack of a simple way to interact CLI-like with the AMI itself. So I have decided to write something myself to make my life easier, or at least a bit less miserable. The result is a little webapp that you can use as a sort of CLI-frontend to the AMI itself. It is not pretty, but pretty much effective. So I thought I could share it and make someone else's life a bit easier. You can find it on https://github.com/l3nz/amiDebugger - if you just want to test-drive it get the WAR file an put it into some webapp container, e.g. Tomcat. Hope you'll like it. l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] looking for a way to do appointment reminders
We did something like that - see http://blog.wombatdialer.com/post/24187267017/drstrangelove You can use the free version of the dialer if you have low traffic or just want to run a test. l. 2013/4/26 Ron Wheeler rwhee...@artifact-software.com Good comment. Another feature suggestion You might to ask the person to press 1 to confirm or 2 to leave a message if the appointment is not going to be kept or 0 to reach the receptionist to reschedule the appointment. Ron On 26/04/2013 7:06 AM, Chris Bagnall wrote: On 26/4/13 10:38 am, jg wrote: they are currently calling patients. I think these calls apply only to a certain fraction of the patients, who are difficult to contact by other methods. I suspect there will be different requirements depending on how 'helpful' to patients you wish to be. At the very simplest end of the scale, you could simply call the patient's number and remind them of their appointment on dd hhmm, then disconnect. However, the OP probably wants something a little more sophisticated than that. At the very least, you would want some method of handling shared numbers (e.g. a shared dwelling with a single phone), so you didn't inadvertently advertise a patient's appointment to someone else who answered the phone. So you would at the very minimum want a simple IVR that says We are trying to reach Mr. Joe Bloggs. If this is he, press 1 now, otherwise please hang up. Going beyond that, you might want your reception staff, when booking appointments, to ask the patient when they would like their reminder call - the day before, an hour before, etc. etc. (and if the day before, would they prefer it in the morning, afternoon, or evening). As others have said, the OP might be best advised to request (paid) assistance with the project on the [asterisk-biz] list. Kind regards, Chris -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
Depends on what you are trying to do. Not in general (AFAIK) but you may find a number of scripts around. 2013/4/18 isr...@gmail.com I think facebook uses xmpp so you could use asterisk jabber or so Don't know about the rest -Original Message- From: bilal ghayyad bilmar...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 17 Apr 2013 14:41:53 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Hello; Is there any modules or channels or integration between asterisk and any of the following: whatsapp, facebook, viber, yahoo and hotmail messanger? Regards Bilal -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Phpagi action based on outbound call user response
I am not sure about PHP AGI, but in general via AGI you can monitor the state of the call and so you can know when the call is over. l. 2013/4/17 Rahul R rahul...@gmail.com Hello List, In PHPAGI, I'm using the Astrisk Manager function send_request() to originate an outbound call. I want to execute the remaining PHP code after the call gets executed (depending on user input). But presently the call originates in a different context and asterisk executes the remaining code in parallel. Is there a way in which I can pause the code execution until the call is completed. Note: I wish to return to the context from which the call was originated and continue execution. Any help is greatly appreciated. -- Thanks Regards Rahul http://about.me/rahulr92 +919567607741 -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] External call control for Asterisk
Not sure if that's what you are looking for, but I would think about having the dialplan call a web service (maybe using CURL) and passing account and current number. The system would reply with the number to actually dial, or none if blocked, and the maximum possible call length. Then it's all Asterisk (or turtles all the way down). 2013/4/10 Simon Green simon.c.gr...@gmail.com Hi there, I’m new to Asterisk and there’s a ton of documentation. I’m not really sure where to start. What I want to do is this: a PBX service ala FreePBX, but where call control is passed via SIP to an external service which will tell Asterisk: a) * Whether the call is allowed b) * Where to connect the call, if necessary (i.e. forced redirection to a C-party) c) * To disconnect the call at some time in future based on charging considerations (i.e. online charging) There is also the option of not using Asterisk at all, and simply using the other service directly, but Asterisk is much better suited to handling end-user devices. The external service does control logic only. Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger
I'd start from https://github.com/venomous0x/WhatsAPI/blob/master/README.mdthat offerts PHP and Java APIS, both not hard to integrate with Asterisk. 2013/4/19 kingman chui chuiking...@yahoo.com.hk Hi, So , how to connect asterisk to whatapps ??Please advice .. Thank Regard/chui king man *寄件人︰* Lenz Emilitri lenz.lo...@gmail.com *收件人︰* isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *傳送日期︰* 2013年04月19日 (週五) 4:34 PM *主題︰* Re: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Depends on what you are trying to do. Not in general (AFAIK) but you may find a number of scripts around. 2013/4/18 isr...@gmail.com I think facebook uses xmpp so you could use asterisk jabber or so Don't know about the rest -Original Message- From: bilal ghayyad bilmar...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 17 Apr 2013 14:41:53 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger Hello; Is there any modules or channels or integration between asterisk and any of the following: whatsapp, facebook, viber, yahoo and hotmail messanger? Regards Bilal -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com/ Test-drive WombatDialer beta @ http://wombatdialer.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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] ACD problem
I am not sure I understand the required routing pattern, but I'm sure queues are your friends, as you can dynamically add and remove member and you can have a first-level queue easily move fall-through to another queue in case all members should be busy or none should be available. Plus by using queues you decouple the what you want to do from the who is doing it. 2013/4/10 Tommy Cooper tomcoope...@yahoo.com Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want to design a system where customers can call my number, that call will then be directed to either extension 1000 or 1001. If both extensions are in use, I want that 3rd call to be queued. I don't think that the config below will direct calls to extension 1001 because the second line states that any incomming calls should be routed to extension 1000. How do I change this so that calls are directed to all of my exensions? extensions.conf [from-myprovider] exten = *DID number*,1,Answer exten = *DID number*,2,Dial(SIP/1000) exten = *DID number*,3,Queue(support) ;not sure if this line belongs here exten = *DID number*,4,Hangup queues.conf [general] [support] musicclass=default strategy=rrmemory joinempty=no leavewhenempty=yes ringinuse=no Member = SIP/1000 Member = SIP/1001 agent = 1000,1000 agent = 1001,1001 When using the current config the caller will listen to the 'music on hold' until the agent answers but calls are only being forwarded to extension 1000 as stated above -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] To queue or not to queue...
Hello Gregory, I wouldn't say this is a typical scenario for using a ringall queue, especially if the agent set gets larger and larger. On the other side, a ringgroup won't solve the issue of ringing all those phones at once. What I would be looking into, considered the motivation of your agents, is to split the system into more than one queue and send the calls randomly to each queue. If everybody is busy you get out and retry. This should not impact call answer times as long as you have 30/40 people available per queue - but your box will handle a fraction of the load and you can easily partition such a system on multiple boxes. Just my two cents, l. 2013/3/28 Gregory Malsack gmals...@coastalacq.com Hello All, History ~ I recently took a position with a call center. At the time they had about 50 agents in a call queue. The queue was setup to ringall. The agents use Eyebeam softphones. Everything is local lan, no routers, everything connected via Cisco 3600 10/100 switches. Now we are up to about 150 agents, and I have kept everything pretty much the same way for a couple of reasons. However, those reasons are slowly drifting away and it's become the right time for me to start questioning some of the previous configuration. Here's the scenario~ 150 agents, all are commission based sales reps. 99% of the calls are answered within the first ring. the rest are answered between the second and third ring. Never in my 4 months with the company has a queue call been in the queue more then 20 seconds. Problem~ Several times a week or sometimes a day, the reps will tell me that the same call will be answered by 3 or 4 or 5 reps, and none of them get the inbound audio. Asterisk only shows 1 of the reps actually connecting the call, however the call logs in Eyebeam for all 5 reps, show that they took the call and were connected for a short period of time before disconnecting the call because there is no inbound audio. Point of discussion~ Is there really a reason to maintain a queue? With the companies growth they are now discussing the option of sending certain affiliates to certain sales reps. Am I better off using ring groups? Additionally I am working towards running as much of my configs via mysql as possible and turning up multiple servers to handle the calls. So far we have reached 130 simultaneous calls on one server, and about 10,000 calls processed during a 12 hour day. Thanks for reading. I look forward to hearing peoples views on this... -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Dedicated LCR Solutions
I know Evariste Systems has a product called CSRP - http://evaristesys.com/pub/CSRP-ProductOverviewCapabilitiesSurvey.pdf - that looks very interesting and it is built for high-volume scenarios. It is basically a standalone box you route calls to. Just my two cents, l. 2013/3/26 Nick Khamis sym...@gmail.com Hello Everyone, Was wondering what some of you for stand alone LCR implementations. I am aware of the LCR module within asterisk and a2billing however, we are looking for a standalone self less coupled solution. Not sure if such thing exist. Kind of like CDR Tool but for LCR... Thanks in Advance, Nick -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Configuration Required for Remove Queue Member
Sounds like the autopause option? l. 2013/1/28 Ahmed Munir ahmedmunir...@gmail.com I would like to know, is there a method in which we can define the timeout value for a member who already login to the queue but after quite a while if he didn't answer the 3-4 calls (not going to member pause queue) but automatically remove the member from the queue? Please advise. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Integration with Social Media, Email and Web call center
I was thinking of something similar, maybe using the URL field of the queue() app as to point to an internal broker that will then link to the message being used. In theory one could do this for all kinds of traffic, including e-mails. The part I don't really like is keeping an audio call open for the duration of the job, but it plays very well with existing queues. In the end, I guess Matt is always right :) l. 2013/1/24 Matt Riddell li...@venturevoip.com In the past I've sent calls to an agent in the queue with music on hold that contained a beep every 20 seconds (to remind them they're on a call) and then used the same code I do for screen popping to send them alternative records. I.E. web page, email, fax etc. It's stored in the database that that's what they were working on and then when they finished working on it they just hang up or press * to disconnect the call. That way you can use the standard Asterisk queues and they don't get bothered by anything else while they're working on it. Facebook might be a little harder as you wouldn't necessarily know when an incoming request came. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Capture queue agent drop and put caller back in queue
2013/1/21 Mitch Claborn mitch...@claborn.net Asterisk 11 Occasionally we will have a partial power outage, or a piece of network equipment will fail, and our queue agents who are on active calls with callers will be disconnected from the caller. What I'd like to do is capture those calls and put them back in the queue (at a high priority) so that we don't lose the caller. I've tried to duplicate the situation in my lab: I have one agent in the queue, a caller dials into the queue, gets connected to the agent then I pull the ethernet cable out of the agent's computer (testing with a softphone) but I don't see anything happen on the asterisk console. core show channels shows the 2 channels still bridged even though the agent is gone. Shouldn't asterisk somehow know when the agent disappears? How can I accomplish my goal? I am not sure that from the PoV of the caller this solution would work - they would experience tens of seconds of silence plus they would have to go back to the queue. If this happens rarely, you could have a process call them back instead - you acknowledge what happened and have someone on-line with the person apologizing. We have a few clients implementing something like this for calls exiting the queue on timeouts and it seems to be well-liked by the callers. Of course it depends on what you are doing and the level of service that callers come to expect. Just my two cents, l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Integration with Social Media, Email and Web call center
And how would you have this working together with Asterisk queueing? I have seen solutions like this using agent pauses and then making everyithing happen outside the normal ACD flow, but it's a bit of a hack l. 2013/1/22 Danny Nicholas da...@debsinc.com For just the messaging part, you should be able to use wget or curl to interface and create messages. You might have to go a little higher level like C or Perl, but it sounds very doable. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, January 22, 2013 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integration with Social Media, Email and Web call center Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? Regards Bilal -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Detect Low Quality Calls - Realtime
2013/1/5 joachim zoach...@securax.org You are pretty much limited to measuring the delay and the jitter. The delay you can somewhat estimate prior to the call (with qualify for example). The jitter / packetloss you can only figure out when the call is already up for a while. (e.g. you might have no issues the first minute, but maybe packet loss will come in bursts after a minute). A few years ago I spoke to a Finnish company that had a commercial solution for automated MOS estimation. So something exists though I have not tested it first-hand. l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] new user help required to build voice recorder with asterisk
I don't think this should be an issue, but we have seen a lot of sites going live and discovering too late that they had recording problems. Maybe you won't need to implement an external recorder, but it's better to plan in advance, not when you are in production! :) l. 2013/1/2 Leandro Dardini ldard...@gmail.com I don't know how many I/O can be achieved on a modern hardware, but I don't think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of data. However can be a good idea to start loading a server and be prepared to share the load on another server. Leandro -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Paging for Praying
How many people do you plan to page? because if numbers are high (or variable) you may have an easier life by using some sort of dialer if numbers are not very high and two lines are enough, our WombatDialer is free to use. l. 2012/12/29 bilal ghayyad bilmar...@yahoo.com 2) Praying time need to be obtained from text (or database). So, it is not always the same time. What actually is needed to be obtained from the text file or the database is the time of the pray for each date (for example, if today is 28 of December so the query will be for this date and then it is required to check if the time is same as the current time to page the wave file on the Phones). -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] new user help required to build voice recorder with asterisk
With just one PRI card this should not be an issue, but for larger systems you may consider using something like Oreka to offload the I/O from the Asterisk server l. 2012/12/31 Vinod Nadiadwala thinw...@gmail.com Hi, I am new to asterisk, i want to know that is it possible to use asterisk for build voice recording system. Scenario : ISDN PRI line (30 line) I want every incoming outgoing call has to recorded, but without manual action. system has to auto receive the call. Please suggest, how should i start and with which hardware / cards it is possible. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Catching hold in dialplan
Steve Murphy submitted a patch a while ago to track MOH on queues, you can find it at https://issues.asterisk.org/jira/browse/ASTERISK-20742 - it could be a good starting point to work on as it is quite short. Too bad it is still in limbo :-( l. 2012/12/19 Andrew White and...@computersforall.com.au Hey all, I’ve built a custom application for our call center and am having one problem. Unfortunately certain things happen whilst the agent has the customer on hold which I’d like to work around. But I can’t work out how to catch the actual hold event so I can do something about it. From the console with verbosity on 12, all I can see is: -- Started music on hold, class 'default', on SIP/trunk-9546 -- Started music on hold, class 'default', on SIP/100-9547 I’m happy to try and catch this AGI or via manager if needed, however a dialplan based solution would be best. Thanks all! Andrew -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Users list email totals by year .
So where has every body else gone? :) l. 2012/12/30 Mr. James W. Laferriere bab...@baby-dragons.com 2003, 24471 2004, 48608 2005, 59116 2006, 41215 2007, 26414 2008, 20746 2009, 18304 2010, 14948 2011, 11588 2012, 7542 -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+ -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] It's possible a redudant Queue?
We have a number of customers who use this approach with local or geographically distributed Asterisks and then use QM clustering to observe the system as if it was one single big box. Seems to work fine and it' easy to set up and maintain. l. 2012/12/14 Danny Nicholas da...@debsinc.com In my experience, you should set up two identical queues and configurations. With a little work, you should be able to let server 1 know the phone is in use by server 2 and vice versa. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi Sent: Friday, December 14, 2012 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] It's possible a redudant Queue? Hi all, I have a doubt. I have to create a queue with 3 phones, these phones can be reached via two redudant Asterisk server. I can pass a variable (the sip trunks) to the queue or should I do two queues with the different trunks? Danilo -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Queue logging
How large is your systems? because the information created by of a call on a queue is just like a hundred bytes, so it is usually safe to keep them all in any case on modern systems. 2012/11/27 Jonas Kellens jonas.kell...@telenet.be Hello, at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information. Is there a way to exclude certain queues from being logged into the queue log ? Thanks, 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Queue_log into MySQL - best practices
Hi Dmitry, we usually advise against writing queue_log events straight to a database, as it is marginally more likely that the DB has issues that a simple flat file. And when data is lost it's lost forever. Still everybody seems to love writing data straight to the DB :) l. 2012/11/22 Dmitry mbike200...@yahoo.com Hi, I use asterisk 1.8. Currently I use a perl daemon to parse queue_log into MySQL. It works reliably. But I know that there is a method ( http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL and http://work.mikeboylan.com/asterisk-queuelog-to-mysql) to write to MySQL directly with app_mysql which has a DEPRECATED status. My question is: What is the best/preffered approach to put queue_log into MySQL in asterisk 1.8 and up? 1) To use external daemons to parse /var/log/queue_log? 2) To use the deprecated app_mysql? the status does not guarantee that this application will be in the future 3) To use odbc to access mysql? but I could not find a procedure for it. And I doubt it is possible. BR, Dmitry Pavlenko -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Astricon 2012 presentations
Thanks - too bad I missed it :) 2012/11/12 Dan Jenkins dan.jenk...@holidayextras.com Hi, As far as I'm aware the videos are still being produced and there's no definitive list anywhere for the slide decks. However, my one is here: http://www.slideshare.net/danjenkins/asterisk-html5-and-nodejs-a-world-of-endless-possibilities-14881614 Dan Jenkins -- Dan Jenkins - Senior Web Developer email: dan.jenk...@holidayextras.com twitter: dan_jenkins http://twitter.com/dan_jenkins linkedin: jenkinsdaniel http://www.linkedin.com/in/jenkinsdaniel skype: d-jenkins blog: www.dan-jenkins.co.uk about.me: about.me/dan_jenkins On 12 November 2012 11:05, Lenz Emilitri lenz.lo...@gmail.com wrote: Hello all, anybody knows if the PDFs for presentations held at Astricon 2012 are available somewhere? I looked at the website but cannot find anything. Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Asterisk 1.8.16 Monitoring tools
Hello Motty, it really depends on what you want to do and the level of detail you want. There are a number of free and commercial applications that can help you in doing this :) l. 2012/11/9 motty.cruz motty.c...@gmail.com Hello, I want to monitor my Asterisk 1.8, inbound, outbound, status calls, queue call? Any suggestions? I found Monast, I'm having issues configurating. Thanks, -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
[asterisk-users] Astricon 2012 presentations
Hello all, anybody knows if the PDFs for presentations held at Astricon 2012 are available somewhere? I looked at the website but cannot find anything. Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] how to lookup a call
I would not know if this is something that can be helpful to you, but in WombatDialer we associate a channel variable to with an unique-id to each call, so that we can reattach to a set of calls if the AMI connection goes down and we can be absolutely sure that what we are looking at is the call we think it is. It is not really expensive to do - just a GetVar per channel to mek sure our assumptions are correct. 2012/11/7 Jerry Geis ge...@pagestation.com I am using 1.4.43 currently. I am using the AMI to originate a call over a SIP Trunk to my cell XXX506. works fine. when the call is active I do a core show channels concise and I get: SIP/testsystem-0ad0!**smvoice-dialout!callprogress!** 4!Up!AGI!smvoice!0!!3!24!(**None) My AGI is called smvoice. No place does my number show up. How do I lookup my call so I can hangup the call at a later time. In my case there my be more than one call active at a time, and I want to hangup the correct call. I know I need the data testsystem-0ad0 to cancel my call but how do I associate that with my number so I can find the right call to hangup. Thanks, Jerry -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Agents in more than one queue at once
In general there is no guaarantee as which call will connect; each queue is independent AFAIK. l. 2012/10/17 Alex Forster a...@alexforster.com My company has been running Asterisk 1.6.2.19-1_centos5 from the official yum repo, and for a while now I've been receiving complaints from our call centers about calls not being routed in the most efficient order. I'll explain with a simplified scenario-- Let's say I have two queues: A and B. I have one agent, Alice, who is a member of both of these queues. While Alice is busy on a call, one person calls in to queue A, and then, several moments later, another person calls in to queue B. At this point, note that both callers waiting on hold are position 1 in their respective queues. A queue show might look like this... A has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s Members: 21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls (last was 533 secs ago) Callers: 1. SIP/Trunk-eb17 (wait: 1:14, prio: 0) B has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s Members: 21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls (last was 533 secs ago) Callers: 1. SIP/Trunk-eb1e (wait: 0:45, prio: 0) My question is: when Alice gets off the phone, which call will she get? My expectation is that she will get the call which has been waiting longer, but I'm not sure that's actually the case. Alex Forster -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Call me now outbound calls in a queue
The problem is that you need to have a process waiting for a free agent and then doing the reschedule. Instead of writing your own, you could try our WombatDialer (that is currently free as in beer, as it is being community tested) to automate such a task. It has a nice HTTP API and it would do exactly what you are looking for. See http://wombatdialer.com/ l. 2012/9/28 Mitch Claborn mitch...@claborn.net That approach only works if there are any agents that are not busy on a call - I could pick one, take them out of the queue then connect the call. If all agents are busy, I need to be able to insert the request into the queue so that it gets processed in sequence with the inbound calls. Mitch -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
The problem I see with this approach is that you usually do not just want to dial out 10 calls at a time, but you will want to keep track of what happened to them and (in case) reschedule them. So you will likely need to monitor them over AMI to make sure they went through, and you need to implement some rescheduling logic. [Shameless plug starts here] This was the reason why we started working on Wombat a while ago - to offer something that would handle all this (and more) but leaving you the Asterisk touch of being free to program the call handling at the dialplan level, so you would get the best of both worlds. Did I already mention the current beta versions are free? :) [Shameless plug ends here] I am not saying that this is the only correct solution (or it is a correct solution at all) but our almost ten years of Asterisk call-center experience show that what starts out as something quick and simple to plug a hole ends up being a platform :) Just my two Swiss cents, l. 2012/9/28 A J Stiles asterisk_l...@earthshod.co.uk On Friday 28 September 2012, Patrick Archibald wrote: Hi, Is there a way to move 100 .call files in to /var/spool/asterisk/outgoing/ at once and have Asterisk call at maximum 10 at a time? Yes: Move them in batches of 10. Could be as simple as last if ++$n_files 9; if the script is in Perl. You know how many calls you can deal with at once; it's up to you to stay within your own limits. Asterisk just tries its damnedest to do whatever it's been told, without imposing any sort of judgement as to whether it's sane or wholesome. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR
Another option that seems to be very good for handling logs where you write quite a lot is Cassandra - http://cassandra.apache.org/ - but of course you lose the SQL layer on top - unless you go for something like http://blog.mariadb.org/announcing-the-cassandra-storage-engine/ This may not be completely off topic here because you get high data security / crash protection and parallel cluster writes, so you could insert tens/hundreds of thousands of events per second on a suitably dimensioned cluster for an Asterisk server cluster of similar size :) l. 2012/9/28 Leif Madsen leif.mad...@asteriskdocs.org On 27/09/12 11:45 AM, Matt Hamilton wrote: Date: Thu, 27 Sep 2012 10:23:35 +0200 From: lenz.lo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR I'd go for MyISAM and would set up a remote replica if data integrity is important. If you have like 1000 calls of (say) 30 seconds avg length, and you create 10 events per call, you would expect an event every three seconds. This is about 300 inserts per second. Say 600 at peaks. This should be feasible with server-grade hardware without much difficulty. Also as you always INSERT it behaves as a log file (no seeking, no locking) if the table is optimized. l. We decided to go with MyISAM since it supports concurrent inserts (as you suggested). Data integrity (a slight loss of call records) is something we can live by. Right now we use DRBD for replication, but I guess with MyISAM it doesn't make much sense if the db crashes. We are looking into other options as well. This may or may not be relevant, but you can also check out MySQL/Galera[0] for clustering solutions. Not sure if that gets you closer or further from your goal though :) It uses a modified InnoDB to allow a multi-master MySQL cluster. I used a chef cookbook to deploy it[1]. [0] http://www.codership.com/content/using-galera-cluster [1] http://support.severalnines.com/entries/21453521-opscode-s-chef-mysql-galera-and-clustercontrol -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] QUEUEHOLDTIME always zero
What do you get if you run a queue show sales? l. 2012/9/26 Mitch Claborn mitch...@claborn.net Asterisk 1.8.10.1~dfsg-1ubuntu1 Trying to build a simple announcement of the queue status. QUEUEHOLDTIME is always zero. What am I doing wrong? queues.conf [general] autofill=yes shared_lastcall=yes [StandardQueue](!) musicclass=default strategy=rrmemory joinempty=no leavewhenempty=yes ringinuse=no announce-frequency = 30 min-announce-frequency = 15 announce-holdtime = yes|no|once announce-position = limit announce-position-limit = 5 announce-round-seconds = 10 setinterfacevar = yes setqueueentryvar = yes setqueuevar = yes [sales](StandardQueue) ; create the sales queue using the parameters in the StandardQueue template extensions.conf exten = 812,1,NoOp(queue status) same =n,Set(LOGGEDIN=${QUEUE_MEMBER(sales,logged)}) same =n,Set(READY=${QUEUE_MEMBER(sales,ready)}) same =n,Set(WAITING=${QUEUE_WAITING_COUNT(sales)}) same =n,Set(STUFF=${QUEUE_VARIABLES(sales)}) same =n,Verbose(waiting: ${WAITING} calls in queue: ${QUEUECALLS} avg hold: ${QUEUEHOLDTIME} logged in: ${LOGGEDIN} ready: ${READY}) Regardless of how long a caller has been waiting in the queue, the output is: -- Executing [812@LocalSets:1] NoOp(SIP/08000F3BE07C-0048, queue status) in new stack -- Executing [812@LocalSets:2] Set(SIP/08000F3BE07C-0048, LOGGEDIN=1) in new stack -- Executing [812@LocalSets:3] Set(SIP/08000F3BE07C-0048, READY=1) in new stack -- Executing [812@LocalSets:4] Set(SIP/08000F3BE07C-0048, WAITING=1) in new stack -- Executing [812@LocalSets:5] Set(SIP/08000F3BE07C-0048, STUFF=0) in new stack -- Executing [812@LocalSets:6] Verbose(SIP/08000F3BE07C-0048, waiting: 1 calls in queue: 1 avg hold: 0 logged in: 1 ready: 1) in new stack waiting: 1 calls in queue: 1 avg hold: 0 logged in: 1 ready: 1 -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR
I'd go for MyISAM and would set up a remote replica if data integrity is important. If you have like 1000 calls of (say) 30 seconds avg length, and you create 10 events per call, you would expect an event every three seconds. This is about 300 inserts per second. Say 600 at peaks. This should be feasible with server-grade hardware without much difficulty. Also as you always INSERT it behaves as a log file (no seeking, no locking) if the table is optimized. l. 2012/9/26 Matt Hamilton mistral9...@hotmail.com Our top priority is the raw Write (INSERT) performance, Read (SELECT) performance is not important. Strict ACID compliance is not necessary either. MySQL (on a separate database server) should be able to handle inserting CDR records (approximately up to 10 records for each call) for about 1000 concurrent calls coming from an Asterisk cluster. Matt -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Help Required IVR
Are you programming the dialplan yourself or are you using a GUI? Also, are you sure that your greetings message is playable by Asterisk? l. 2012/9/24 Farooq Hussain farooqhussain...@gmail.com Hello everyone, I stuck in problem I have creating a time based IVR and its working fine. If my IVR playing in office hour it would standard IVR and if not they we have play a greeting message and place that call to voice mail of a extension. My problem is this I am able to transfer the call on voice mail but how to play greeting message first. I am using trixbox 2.2.8 anyone help is this regard would great full. -- Thanks Farooq Hussain -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] BLF and Call Queues
In general I would not use this for a true call-center with hundreds of agents, where it is the ACD's responsibility to route calls to agents and there are strict policies on agent behavior, but I'm sure there are a number of cases where this could be useful (eg small call centers, internal service desks, receptionists, etc...). Just my two cents, l. 2012/8/21 Olivier oza_4...@yahoo.fr Hi, What about Queue logs ? How is a picked-up call logged ? Giving agents the capability to easily pickup a call, without beeing logged-in, is a big change with both positive and negative side effects. I would be curious to read opinions about that. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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
Re: [asterisk-users] Need queue name in CDR
It would likely be easier for you to use a tool that already processes queue_log information. There are a number available :) 2012/6/13 Pratik Shrestha pratik...@gmail.com Dear All, I am making asterisk report using CDR values given by asterisk. I have queues which consist of multiple members (extension). Also, an extension may be in multiple queues. So, I want CDR to record the name/number of queue from which the call was originated. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer alpha @ http://wombatdialer.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
Re: [asterisk-users] Asterisk Capacity
We are working on a project to create a general-purpose telecasting server - see http://wombatdialer.com - there is practically no documentation yet, but it's easy to set up and we tested it originating hundreds of channels on multiple servers. It is alpha stage, but current versions are free and I expect them to basically work. If you want to give it a shot, you can install via RPM as described on the website. Thanks l. 2012/5/3 Ashish Agarwal ashisha...@gmail.com So what is a better approach to achieve this On May 3, 2012 9:20 PM, Mitul Limbani mi...@enterux.in wrote: The other 70 will result into failure with .call file approach. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal ashisha...@gmail.comwrote: Hello, We are currently working on a project where using .call file on asterisk spool, outbound calls will be made from a pri line and a voice clip will be played. We know that pri has a capacity of handling only 30 channels at a time. Therefore, my worry is what happens if we write 100 files at a time on the spool. Will asterisk manage the queue or how exactly will it behave. Regards, Ashish -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Asterisk Capacity
At the moment it's free as in beer, though closed-source. It is written in Java and uses MySQL as its back-end. l. 2012/5/11 Arstan arst...@gmail.com Hi, wombat looks promising. Questions: What technologies are used? Is it open source license? On Fri, May 11, 2012 at 2:26 PM, Lenz Emilitri lenz.lo...@gmail.comwrote: We are working on a project to create a general-purpose telecasting server - see http://wombatdialer.com - there is practically no documentation yet, but it's easy to set up and we tested it originating hundreds of channels on multiple servers. It is alpha stage, but current versions are free and I expect them to basically work. If you want to give it a shot, you can install via RPM as described on the website. Thanks l. 2012/5/3 Ashish Agarwal ashisha...@gmail.com So what is a better approach to achieve this On May 3, 2012 9:20 PM, Mitul Limbani mi...@enterux.in wrote: The other 70 will result into failure with .call file approach. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Thu, May 3, 2012 at 9:11 PM, Ashish Agarwal ashisha...@gmail.comwrote: Hello, We are currently working on a project where using .call file on asterisk spool, outbound calls will be made from a pri line and a voice clip will be played. We know that pri has a capacity of handling only 30 channels at a time. Therefore, my worry is what happens if we write 100 files at a time on the spool. Will asterisk manage the queue or how exactly will it behave. Regards, Ashish -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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 -- Regards, Arstan Jusupov -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Official numbering plan - where to get?
I had a look at the files and they are really a nightmare to parse. Some are Word and some are Excel. Good luck :) l. 2012/3/29 Markus unive...@truemetal.org http://www.itu.int/oth/T0202.aspx?parent=T0202 But don't do it. Because I'm doing it right now. So let's not waste energy and do the same task twice. A complete list will soon be available, for free. And then we on this list here will start a web project to keep it updated. I'll let you know once the list is ready. I've already registered opennumberingplan.org :) Am 29.03.2012 11:12, schrieb Lenz Emilitri: DO you know if the doc files from the ITU are available somewhere for download? l. 2012/3/22 Markus unive...@truemetal.org mailto:unive...@truemetal.org I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com http://numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Official numbering plan - where to get?
DO you know if the doc files from the ITU are available somewhere for download? l. 2012/3/22 Markus unive...@truemetal.org I hope this is not too off-topic. As a kind-of follow up to rate sheet normalization I'm still trying to figure out a solution for: throw 10 ratesheets at a program and get the cheapest codes/providers as output. For this purpose I believe I need a real, detailed, accurate list of all the dialing codes, incl. mobile codes, city codes etc. worldwide as a reference for that particular program. There are thousands of A-Z lists on the web, and there are thousands of codes in them, but nothing is accurate enough or from an official source. So, I spoke with the ITU today and they, funny enough, too don't have such a list. At least they don't have one that is computer parseable, like a .csv or .xls or something like that. What they have is: a single .doc or .pdf file for EACH country (1 file per country), which is not standardized in its content, with lots of text and descriptions, but it has all the codes. They don't even have such a list as a paid service. Feels like 30 years ago. :) Anyway, there is numberingplans.com which provide exactly what I'm looking for, but they don't support one-time purchases, only subscriptions from around 100 to 990 EUR per month, which is above my budget (and I don't need a subscription). Does anyone have an idea where to find such a list for free, or as a one-time purchase? If not, I'll probably go through the effort to compile my own list based on the ITU data. Let me know in case you want a copy then. :) -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Collaboration Call Center Integrated with Asterisk web and email
A number of call-centers I see use the pause codes in Asterisk to mark different types of activities, like answering to email or IM. It's not much, but easy to implement. l. 2012/3/27 bilal ghayyad bilmar...@yahoo.com Hi All; Is there a collaboration contact center (hope to be open source) Integrated with Asterisk (hope with vicidial), so the agent will be able to receive chat or emails sessions and deal with the customer. If the agent in a call with the customer, then he will not get chat session. Is there like this software? Regards Bilal -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Asterisk 1..8 multiple queue
Shameless plug: the QueueMetrics agent page, even in the free 2-agent version, can emulate this behavior. You may want to check it out. l. 2011/5/25 satish patel satish...@hotmail.com Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Asterisk Queue ACD when the queues and agents has the same priority/weight
Each queue is separate and does not see what other queues are doing. l. 2011/3/23 Marcos Setim m.se...@gmail.com Hello, I have three queues (F1,F2,F3) with default queue weight and three agents (A1,A2,A2) with default agent penalty. If the three agents are busy and tt same time a caller (C1) enter in the queue F1, and after 20 seconds a second caller (C2) enter in the queue F2. So, few seconds later, the agent (A1) state comes to availabe. In this case the asterisk deliveries the caller (C2) to agent (A1), but the in the queue (F1) caller (C1) waiting time is bigger compared to caller (C2) of queue (F2). How should be the ACD behavior between queues in this case? How the asterisk distributes incoming calls when the queues and agents are the same weight/penalty? Thanks, -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Queue pause vs logged out ?
Maybe not much from the point of view of queues, but this may make quite a difference from the point of view of monitoring your call-center. :) l. 2011/3/21 satish patel satish...@hotmail.com Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -- -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] How to reject an incoming call using AMI ?
Why not an unattended transfer to the queue itself, or a different queue? l. 2011/1/10 Olivier oza_4...@yahoo.fr Hi, For a call center, I'm studying how I can offer agents the ability to reject an incoming call using a custom application. As you can guess, in this case, rejecting a call means let another agent answer this call (it doesn't mean end this call). The only way I could imagine for this to happen, would be to redirect the caller to a conference room, then hangup the agent call leg and then redirect the caller back to the appropriate queue, hoping the caller wouldn't be once again forwarded to the busy agent. Which way to implement this would you suggest or recommend ? Regards -- Loway - home of QueueMetrics - http://queuemetrics.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
[asterisk-users] New tutorial: Compiling Asterisk 1.8 on CentOS 64
Hello all, as everybody else here - I guess - I have been playing with the new Asterisk 1.8 release. So far everything went smoothly - the compilation phase was really straightforward, and I have a box ready for real testing now. I prepared a tutorial out of my experience on how to compile Asterisk 1.8 with iCal, GTalk, SNMP, MySQL, cURL and DAHDI - the usual stuff - so if anybody is interested or has suggestions/improvements, it's here: http://astrecipes.net/index.php?n=398 .I did not include H323 this time as I don't have H.323 gear anymore to test it with! :) Comments are welcome. l. -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Queue member status - BUSY
Have you tried playing with joinempty and leavewhenemèpty to avoid people being connected to a queue with all agents in use? l. 2010/10/20 GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.com Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas? regards, ryan icasiano -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Solving the CDR mess of attended transfers
Is there a documentation about the CEL format? l. 2010/9/22 Steve Murphy m...@parsetree.com CEL was my answer, built on the channel event goodness that Russell. It's now in 1.8; but it lacks a converter to CDRs. You *could* just use the string of events coming out of CEL, but... I'd love to see your SQL statements to pull things together! I had begun writing a CEL-CDR converter, but got laid off before I could finish it. It makes a good start toward a finished package. Long ago (what, almost 2 years now?) I proposed two methods of generating CDR's. One was 'simple', the other 'Complex, or Leg Based. Since then, I refined the doc to just 'Simple', and outlined with some examples how it would/should work. The doc still needs to be cleaned up, but you may make sense of it. The trouble with CDRs is that no two shops can agree on a CDR standard that involves transfers, parks, etc. Beyond the start, answer, and end times, and some fundamental data about the call (source, dest, responsible party, etc.) There isn't much unity about what timepoints need to be represented, etc. And I'd seen so few implementations, I couldn't judge a good way to generalize the CDR converter. So, I challenge everyone to look at my simple CDR definition, and see it would possible for you to adapt it (perhaps via a mapping from it to your desired conflagration/configuration. To look at the doc, do svn co http://svn.digium.com/svn/team/murf/asterisk-RFCs and look at the document in there (I have a few different formats, the .docx is the source). It's been in flux. Just the first few examples are accurate. Let me know what you think. murf -- Steve Murphy ParseTree Corp -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] How to track a call result originated from originate AMI command
That was exactly what I was lamenting - that some common distros do not send every event, so that AMI ends up being less than reliable. If AMi sends all events, then it's really trivial to track calls :) l. 2010/8/9 Motiejus Jakštys desired@gmail.com On Mon, Aug 9, 2010 at 12:08 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: BTW, using the most common Asterisk distros out there that happen to sport a very complex dialplan, we see a lot of lost events, so that tracking calls on the basis of AMI observation alone becomes practically impossible. :-( l. You can filter AMI. If you know PERL, you can start with my script that works with callbacks: $callbacks{'Newstate'} = \newstate_callback; $callbacks{'Dial'} = \dial_callback; And create appropriate functions for storing desired values to the database. We catch Dial, Answer, Ringing, Hangup events and store that info to database with very accurate timestamps :-) http://github.com/Motiejus/Asterisk-perl-AMI/blob/master/asterisk_ami.pl Regards, Motiejus Jakštys -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] How to track a call result originated from originate AMI command
BTW, using the most common Asterisk distros out there that happen to sport a very complex dialplan, we see a lot of lost events, so that tracking calls on the basis of AMI observation alone becomes practically impossible. :-( l. 2010/8/8 Nasir Iqbal na...@ictinnovations.com Hi, Confusing! you are not alone here. Actually there is no unified development approach exist in Asterisk, every module, application introduce a new way to handle same things!! And the monitoring is most difficult part! you have to write different parsing algos to get each bit of information, and unfortunately you have to rewrite most of your code for every new release! And regarding your question, I recommend you to use AGI for monitoring here is some tips for you - in originate command use extension as destination. - create failed extension in same context. - you can include some variables in originate command which can be used later in dialplan. - use AGI scripts in destination and failed extensions to get and save call status in database. Regards -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
Well, actually we are in contact with quite a number of call-centers that use the free version - a lot of times it's embedded call centers, like internal help-desks and such. One of the nicest things of * is that you would not buy an ACD module for a traditional pbx to support just a couple of users, but with * it's free. l. 2010/7/31 bruce bruce bruceb...@gmail.com 2 users. So, it's probably never used as a free version as probably there are no 2 seat call centers that can survive this economy. But, it should great for testing. On Sat, Jul 31, 2010 at 10:28 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 7/30/2010 5:49 AM, Lenz Emilitri wrote: QueueMetrics is actually free (as in beer) for very small call centers and individual hackers. Oh really! I didn't know that! Very nice. What is considered a small call centre? Are we talking up to around 5 agents or something? Is there a limit on the number of queues? (I'm sure there is a page on the website that answers most of these questions, heh :)) Leif Madsen. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
QueueMetrics is actually free (as in beer) for very small call centers and individual hackers. l. 2010/7/28 Zeeshan Zakaria zisha...@gmail.com There is none for free. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-27 6:12 PM, bruce bruce bruceb...@gmail.com wrote: :-) I knew someone would bring up FreePBX. I have FreePBX installed and it's not good for Queues at all. It's using the reporting tool from Areski and Areski has recently released an upgrade to it which again is not what I want. There are few other programs that do this but really none that are neat in interface or useful in features. I guess no one else has any thoughts on this? Maybe there is none available? Thanks, Bruce On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg dbackeb...@gmail.com wrote: On Mon, Jul 26... -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Asterisk distribution for a Call Center
It really depends on how large your CC will be and how much money is at stake. :-) We have a lot of clients who are very satisfied with small call centers based on FreePBX or Trixbox CE. Of course I would not implement a 500-seats call center out of a standard CD. My suggestion is: make sure you have an experienced local consultant handy in case something goes wrong - in real life, it always does. Just my two eurocents, l. 2010/6/22 Alejandro Cabrera Obed aco1...@gmail.com Dear all, I need to build a PBX based on Asterisk for a call center. I have worked with raw Asterisk but it's hard to work for big implementations think. Also I have worked with Trixbox CE for a small bussines and it was prette good, but I have not have many features like ACD. I know there is another version called Trixbox PRO -specially Call Center edition- that's not free but has got more features like ACD and billing. I've heart about AsteriskNow and I know it's free. What distribution/version do you recommend to me in order to implement a call center and taking into account I'm not an expert in programming from Asterisk CLI ??? Thanks a lot Alejandro -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Agents
Use Addmember and removemeber instead :) l. 2010/5/14 Peter Childs pchi...@bcs.org I've been trying to get the hang of Agents and Queues and I must say its a little unclear as to how things work. So maybe someone has some better idea From what I can work out an Agent is meant to be nothing more than a virtual device that can be moved around physical devices... by login and logging out. Queues can contain any type of interface not a point that is partially well put in the Sark we have just got nore in the voi-info website It also seams to suggest that Agents are a deprecated feature. AgentLogin. AgentCallbackLogin is depreciated but what has it been replaced by? Not sure what AgentLogin is actually useful for. AgentCallbackLogin in the Management API does not set ${AGENTBYCALLERID_${CALLERID(num)}} I guessing this is a error, fortunatly I've worked out a way to get round it. (setvar) The is no way to log an agent in from the Command Line Interface. AgentLogoff Easy so long as you know the agent id you need to logoff, which means using ${AGENTBYCALLID_${CALLERID(num)}} Queues really have very little to do with Agents as any type of device can be statically on a queue or dynamically added when needed, but the info I've found seams to heavily tie the two concepts together. Peter -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Problem with callerid(dnid) and queue
You sure it's not using the URL OPEN parameter for the very queue? l. 2010/5/11 Carlo Dimaggio jaasmail...@gmail.com Hi all, In order to use the open url function of zoiper (it opens an url based on the asterisk $callerid(dnid)), I need rewriting of the dnid. In my dialplan I have: exten = 1000,3,Set(CALLERID(dnid)=newdnid) exten = 1000,4,Noop(${CALLERID(dnid)}) exten = 1000,5,Queue(test-queue) but the callerid(dnid) shows the extension called (the member of the test-queue) and not the newdid. I have tried also with the option o in cmd Dial but without success. Do you know if there is a way to obtain the newdnid? Thanks! Carlo -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
[asterisk-users] Check if extension loaded over AMI
Hello list, I was wondering if there is a way to see if a given piece of dialplan is loaded through AMI. I have seen the GetConfig command, but it seems to expect a file name to retrieve, and I don't necessarily know that (as it could be down the line bu multiple levels of #includes from the main extensions.conf). I could run an AMI Command to run the cli command dialplan show mycontext, but I'm a bit worried by the performance cost of running a non-natively AMI command; plus I don't love much the line-formatted response. I could create a dummy piece of dialplan that is in the same place as . the one i want to check, and I could try and Originate that and see if it found or not. All solutions above seem to be suboptimal any idea? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Evaluating Asterisk
Hello Ted, feel free to contact us off-list - we have quite a number, from smallish to extremely large, with varying degrees of clustering and redundancy, in nearly any country in the world! :) l. 2010/4/19 Ted Foote t...@abscollect.com I am thinking of moving from a traditional PBX to an asterisk box. Many of my leadership group are skeptical of asterisk. So I was hoping to find a call center that is currently using this technology that would not mind spending some time on a conference call to address some concerns that my team has. Thanks Ted Foote Allied Business Services, Inc. 616-741-0437 -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Queue call to specific queuemember
Using multiple queues? l. 2010/4/15 Asterisk Maniac asteran...@gmail.com Hi all, What would be the best way to send a call to a queue as usual, but telling that it should be awsered by some specific member? Thanks already -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8
Just can't wait for the live calorie counter! :) l. 2010/4/1 Olle E. Johansson o...@edvina.net FOR IMMEDIATE RELEASE Puerto Escondido, Mexico, April 1st, 2010: Digium launches Asterisk VCC (TM) - a new virtual communication platform for enterprises, the public sector and the home. === For VoxSwitch customers, VCCnet will mean that every user can monitor the movement of coworkers in realtime. By using the new APIs, additional data like credit card transactions, fuel consumption in the car, mileage in the air and calories eaten can be reported with a 3D graphical display using HTML5. -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] Asterisk system for church call center
We have a lot of clients who run small call centers based on Trixbox, and seem to be pretty happy with them. Have a look here: http://queuemetrics.com/manuals/QM_Trixbox-chunked/ Thanks l. 2010/3/31 Frank Church voi...@googlemail.com On 29 March 2010 21:46, Frank Church voi...@googlemail.com wrote: I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5 agents, exactly as they have described it. 1. Take incoming calls 2. Put them on hold if there is no one to handle the call immediately, or transfer them to an available agent 3. Take down their details, and number, (if this can be retrieved and saved from the caller id, thats better) 4. Get them to hold on after taking their details if they still want to hold 5. Call them back when the backlog is cleared up. I have a fairly good grasp of the hardware and programming part of Asterisk, having compiled it more than a few times and implemented A2Billing phone card and call shop system with it. But the type of software suited to the Call Center side is where my knowledge gap lies. I am looking for solutions based on the usual Asterisk distributions like AsteriskNow, trixbox, elastix etc, whether ready packaged or requiring additional customization. The matter of whether they will use soft phones, or regular phones with headsets is also something to consider. Soft phones with good GUI's may be preferred if more cost effective for them, although my personal preferences are with hard phones. Any recommendations - the ease of software for the end users is the main thing for me, and integration with the database for taking customers details is the main thing for me. One of the distributions with SugarCRM comes to mind here. Sorry for cross-posting, but ready made and commercially supported systems are not ruled out, if they come within their budget. Regards Frank Church After there response I will go with some of ready made Asterisk distributions, then consider to go for a commercial supported versions if they do not meet the churches needs. Thanks Frank -- Loway - home of QueueMetrics - http://queuemetrics.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
[asterisk-users] Looping over AstDB
Hello list, anybody has handy an example of how to loop over an ASTDB family by getting all the keys in the dialplan? Like I have the AstDB set as: /test/102 : 205 /test/106 : 203 /test/113 : 209 I would like to get (in any order) the 102, 106 and 113 as members of the family test. TIA, l. -- Loway - home of QueueMetrics - http://queuemetrics.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] Important security alert: update your dialplans now!
Yes that's cool! :) l. 2010/2/17 Miguel Molina mmol...@millenium.com.co Ok, if I get it the simplest workaround would be changing this: exten = _X.,1,Dial(SIP/${EXTEN}) To this: exten = _X.,1,Dial(SIP/${FILTER(0123456789,${EXTEN})}) If you're intended to receive only numbers from the dialstring, right? See http://www.voip-info.org/wiki/view/Asterisk+func+filter Regards, -- Loway - home of QueueMetrics - http://queuemetrics.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] Important security alert: update your dialplans now!
Ok but this is available today and works fine, so it can be used as a zero day replacement. Any syntax change is welcome but will take time until it gets in a public release and does not save you the hassle to change the dialplans anyway - unless you implement it as a default behaviour at the SIP driver level. And I got a feeling that most people will simply not bother learning regexps You could just as reasonably write a script to do the check, or run a check in the dialplan itself, or change Asterisk. l. 2010/2/15 Steve Murphy m...@parsetree.com On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri lenz.lo...@gmail.comwrote: Yes but in any case you can enter all of the strings that reasonably match - even if you have variable-length numbers, you will be able to determine that a valid number be between 5 and 15 characters - or likely 2 to 20, all numbers. A number of 156 characters is very likely to be a problem. This is probably a stupid idea, because it could only be implemented in trunk, and won't help with current implementations, and I suggested it a long time ago already when I did the fast pattern matching code, but I don't THINK it would be all that hard to offer SOME regex syntax in patterns to help reduce the impact of these kinds of problems. -- Loway - home of QueueMetrics - http://queuemetrics.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] Important security alert: update your dialplans now!
Or one could simply rewrite to: [incoming-from-voip] exten = XXX,1,Dial(${ext...@incoming-from-voip-old) exten = ,1,Dial(${ext...@incoming-from-voip-old) exten = X,1,Dial(${ext...@incoming-from-voip-old) exten = XX,1,Dial(${ext...@incoming-from-voip-old) [incoming-from-voip-old] exten = _X., 1, dial(SIP/${EXTEN}) To avoid extensive rewriting and fix the current issue. l. 2010/2/14 Steve Edwards asterisk@sedwards.com On Sun, 14 Feb 2010, Kyle Kienapfel wrote: strip_ampersands(${EXTEN})? (sip.conf) [general] allow-characters= all disallow-characters = [example-did-provider] allow-characters= [:numeric:] - -- Loway - home of QueueMetrics - http://queuemetrics.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] Important security alert: update your dialplans now!
Yes but in any case you can enter all of the strings that reasonably match - even if you have variable-length numbers, you will be able to determine that a valid number be between 5 and 15 characters - or likely 2 to 20, all numbers. A number of 156 characters is very likely to be a problem. BTW, you could add a catchall mail the sysadmin option - so when you get a number that is not being matched you could be notified and adjust the dialplan as needed. l. 2010/2/15 Olle E. Johansson o...@edvina.net To avoid extensive rewriting and fix the current issue. That works in countries where you have fixed-length numbers. Unfortunately, not every dialplan works that way, so that can't be a generic advice even though it may solve your problems. Thanks for your suggestion! /O -- Loway - home of QueueMetrics - http://queuemetrics.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] Can an agent Login to a queue and be paused
In this case, I suggest you modify the login script so that your agents always start paused. It should be trivial to do. l. 2010/2/8 Robert Grignon rgrig...@fleetone.com Not a bad idea... We use queuemetrics and the login is done via Web GUI. I could easily just send it to pause upon login... -- Loway - home of QueueMetrics - http://queuemetrics.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] Can an agent Login to a queue and be paused
I'm not sure if this works for newer versions of Asterisk, but on old ones, you could pause an agent and THEN log him on, and he'd be paused. l. 2010/2/4 Robert Grignon rgrig...@fleetone.com I thought there was an option for this but cant find it We have a busy callcenter and I would like the agents to log in and be in a paused state upon login... Right now they login and they are instantly receiving a call Thanks for the input... -- Loway - home of QueueMetrics - http://queuemetrics.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