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We have Elastix version 2.4 setup and we have some little questions which if you could please answer to solve via web based panel of Elastix, as agents are making calls via Elastix there are little issues we are having, like:

  1. Agents make very long duration calls, so we want to setup a rule or whatever possible in Elastix to terminate all calls(for every agent) after 20 minutes of duration, and for agents who comes in evening shift after 6:00 to terminate their calls after every 30 minutes.

  2. Can we except any particular agent from above rule? like we know someone who makes all legitimate calls without any issue so if we need to exempt him can we do this?

  3. How can we switch/block the country based code according to service provider, like we have one service provider which is costing too much for Nigeria (code 00234), so any call going to this code should automatically be routed to other service provider which we have in trunk which costs cheaper than first one. Someone told me this is done via outbound routes in the Elastix application? if that is the case can someone please tell me the steps from exactly where can I do this? and any simple example rule?

Please note that we have access to Elastix via web based panel and if you could kindly let me know for these to perform it from there?

Regards, Fari

Fari
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1 Answers1

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1) can be done by changing "Asterisk Dial Options" in "Advanced features" tab. You need add S(1200)

S(x): Hang up the call seconds after the called party has answered the call.

2) no, you can't in default elastic install. However you can request expert do module for such task( need add into sip_additional.conf variables and use that variables instead of S). Or use custom context module(also require advanced asterisk knowledge).

3) It can be done like this:

First you need create trunk which always answer call and say something like "blocked". That can be done by using announcment,fake inbound did (for example 1111) and custom trunk with dialstring Local/1111@from-pstn.

After that you have put in outbound routes all codes you not want pass to that trunk. As result such codes will go your fake trunk->fake did->anounce.

Other posible option is use custom dialplan with fake trunk like above.

arheops
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  • Hi, The version we have of Elastix is 2.4 and I don't see any "Advanced features" tab where I can edit "Asterisk Dial Options", Yes, but I see "Asterisk Outbound Dial command options" where we have already specified L(1200000) which is disconnecting call's after every 20 minutes...but issue is that it is happening for both morning and evening shifts while we need call termination for morning 9 to 5 for 10 minutes and rest of the times it should be 20 minutes. How can we achieve this? – Fari Nov 29 '14 at 12:55
  • You can ask someone change it twice per day ;) or you can ask someone write module for your shift. No, you can't do that in freepbx/elastix using web only. – arheops Nov 29 '14 at 20:44
  • Hi, 1. Are you aware of such module which I can see? and edit it according to our requirements? 2. While blocking calls via routing them to fake trunk, we want to play a recording message like "This call is blocked due to company policy reasons...." how can we do this? can you please explain as I am newbie :) – Fari Dec 01 '14 at 09:42
  • There are no such module. But not a problem write one if you have skilled enought programmer. I described above how to play message. Just record message, put it as announce and route fake trunk(dial line already described in prevous message) to announce. – arheops Dec 01 '14 at 14:50
  • Hi, Ok, so when you say "Just record message" I will do it via Windows recording tool as WAV file, "put it as announce" how to do this? and where actually? "(dial line already described in prevous message) to announce." Can you please define as I am struggling to understand it, and can you tell where to do this in Elastix web panel.. what only I am doing at this time is just moving all the not allowed calls to fake trunk nothing else. – Fari Dec 03 '14 at 07:59
  • It is short questions site, not howto manual one. Read doc at freepbx.org web site. – arheops Dec 03 '14 at 08:06