[Amportal-devel] Amportal-devel Digest, Vol 10, Issue 1

  1. IVR message after selection (ip-rob)

to achieve what you want in todays’s setup:

IVR -> Annoucement(module) -> Queue

  1. Programically reload (avoid the red bar) (stevesy)

you want ‘clustering’ which is on the bounty page. Your request is a specific feature within such a module. To do it correctly involves a central gui for everything. It’s on the bounty page - feel free to make a pledge towards developing it. http://www.freepbx.org/bounties/

  1. Intercom Auto-Answer capability with Aastra 480i IP Phones…

We plan on modifying the paging module to accomodate such customizations and eventually devices will have it. It will apply to both intercom and paging. If you would like it accelerated let me know how much you might be willing to contribute. Otherwise you can just patch the feature code for now that hard codes the string to what you need. (if you want to pay for a patch for you to do it I could also do that  - you can reach me on the IRC #freepbx or #freepbx-dev (where I pay more attention).

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Today’s Topics:

  1. IVR message after selection (ip-rob)
  2. Re: bosRX (Abe Schreiner)
  3. Programically reload (avoid the red bar) (stevesy)
  4. Programically reload (avoid the red bar) [Edited] (stevesy)
  5. Intercom Auto-Answer capability with Aastra 480i IP Phones…
    (KMorley)
  6. Intercom Auto-Answer capability with Aastra 480i IP Phones…
    [Edited] (KMorley)
  7. [new feature] assistance with incorporating new feature (ip-rob)
  8. Re: [new feature] assistance with incorporating new feature
    (ip-rob)
  9. Re: [new feature] assistance with incorporating new feature
    (ip-rob)
  10. mISDN and freepbx (Christian)
  11. hi (Mihammad Ligon)

Message: 1
Date: Sat, 27 Jan 2007 01:03:05 +1000
From: ip-rob
Subject: [Amportal-devel] IVR message after selection
To: [email protected]
Message-ID: [email protected]

We use the IVR to route people to a support queue. We also have a DID set up for the queue so that people can dial it directly. I’d love to see a feature on the IVR configuration that allows a “post-selection” message. Today we tell the direct number by configuring a join message to the queue. It is a bit annoying to people since they get that message even if they dial the direct number. I’d like to configure the message to be played at the IVR level after the selection.

It seems like the way this would be implemented in the dialplan is to add a playback just before the goto statement. I’d modify the extensions_additional.conf but it will just be overwritten.

I’m sure the ivr table may have to be modified as well to include a flag on whether to have a playback or not…and then the function that writes out the ivr portion of the conf file.

Comments? I could take a stab at some of the modifications with a little help.

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-------------------- m2f --------------------


Message: 2
Date: Sun, 28 Jan 2007 08:38:54 +0100
From: "Abe Schreiner"
Subject: Re: [Amportal-devel] bosRX
To: "Dodie Ronco"
Message-ID: 01c742af$5ca4e950$0301a8c0@audeleyukbcdyp
Content-Type: text/plain; charset=“us-ascii”

Hi,

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Remove “*” to make the link working!

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crying with exasperation.
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Message: 3
Date: Wed, 31 Jan 2007 12:56:16 +1000
From: stevesy
Subject: [Amportal-devel] Programically reload (avoid the red bar)
To: [email protected]
Message-ID: [email protected]

I’ve seen this requested several times and the answer to this is write your own system.

I personally love FreePBX, I don’t want to write my own. The GUI is not what makes FreePBX.

Here’s the reason I want to bypass the evil red bar. I have several sites that are trunked together. We have emergency failover if a PRI dies. All numbers from that PRI are automagically routed to one of the other locations. What we do is we route calls to an IAX trunk to the appropriate box if the incoming call does not belong on the PRI.

For instance:

Box1 has dids 19992220000-19992220099
Box2 has dids 19993331000-19993331099
Box3 has dids 19994442000-19994441099

IAX Trunks:
Box1 to Box2, Box3
Box2 to Box1, Box3

etc

19993331000 comes in on box1 and it’s routed to box2 through the IAX trunk because the DID exists on Box2. This allows for complete failover of PRIs.

In any case we have a lot of numbers to deal with. Every time we add a DID we have to add it to all three (actually more) boxes. We want to do this through a central location. We’ll write the front end but we need to know how to programically reload each box without having to manually login and click the red bar on every box.

Sorry for the long explanation just want to make it obvious why we need this feature.

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Message: 4
Date: Wed, 31 Jan 2007 13:05:18 +1000
From: stevesy
Subject: [Amportal-devel] Programically reload (avoid the red bar)
[Edited]
To: [email protected]
Message-ID: [email protected]

This is an edited version of a previous post

I’ve seen this requested several times and the answer to this is write your own system.

I personally love FreePBX, I don’t want to write my own. The GUI is not what makes FreePBX.

Here’s the reason I want to bypass the evil red bar. I have several sites that are trunked together. We have emergency failover if a PRI dies. All numbers from that PRI are automagically routed to one of the other locations. What we do is we route calls to an IAX trunk to the appropriate box if the incoming call does not belong on the PRI.

For instance:

Box1 has dids 19992220000-19992220099
Box2 has dids 19993331000-19993331099
Box3 has dids 19994442000-19994441099

IAX Trunks:
Box1 to Box2, Box3
Box2 to Box1, Box3

etc

19993331000 comes in on box1 and it’s routed to box2 through the IAX trunk because the DID exists on Box2. This allows for complete failover of PRIs.

In any case we have a lot of numbers to deal with. Every time we add a DID we have to add it to all three (actually more) boxes. We want to do this through a central location. We’ll write the front end but we need to know how to programically reload each box without having to manually login and click the red bar on every box.

Sorry for the long explanation just want to make it obvious why we need this feature. I’ll even pay someone to make a perl script for me as long as its reasonable.

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Message: 5
Date: Thu, 1 Feb 2007 05:04:57 +1000
From: KMorley
Subject: [Amportal-devel] Intercom Auto-Answer capability with Aastra
480i IP Phones…
To: [email protected]
Message-ID: [email protected]

This refers to FreePBX 2.2.0, Asterisk 1.2.14 and Aastra 480i firmware 1.4.1.1077.

FreePBX auto-generates extensions_additional.conf with the [ext-intercom] context containing a workable hands-free intercom section.

The problem is that the stock [ext-intercom] context won’t trigger the auto-answer-alert splash tone on Aastra 480i phones using firmware 1.4.1.1077. That firmware requires a special SIP header to activate the splash tone.

To make it work, you have to REPLACE line 7:

exten => _*80.,n,Set(__ALERT_INFO=Ring Answer)

WITH:

exten => _*80.,n,Set(__SIPADDHEADER=Alert-Info: ;info=alert-autoanswer)

Unfortunately, every time that you use FreePBX to maintain extensions, it overwrites the changes.

I would like to know where in the FreePBX code this context is generated. I might be able to modify my copy of FreePBX, if I could just find the right module and the code.

Thanks in advance for any suggestions!

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Message: 6
Date: Thu, 1 Feb 2007 05:05:14 +1000
From: KMorley
Subject: [Amportal-devel] Intercom Auto-Answer capability with Aastra
480i IP Phones… [Edited]
To: [email protected]
Message-ID: [email protected]

This is an edited version of a previous post

This refers to FreePBX 2.2.0, Asterisk 1.2.14 and Aastra 480i firmware 1.4.1.1077.

FreePBX auto-generates extensions_additional.conf with the [ext-intercom] context containing a workable hands-free intercom section.

The problem is that the stock [ext-intercom] context won’t trigger the auto-answer-alert splash tone on Aastra 480i phones using firmware 1.4.1.1077. That firmware requires a special SIP header to activate the splash tone.

To make it work, you have to REPLACE line 7:

exten => _*80.,n,Set(__ALERT_INFO=Ring Answer)

WITH:

exten => _*80.,n,Set(__SIPADDHEADER=Alert-Info: ;info=alert-autoanswer)

Unfortunately, every time that you use FreePBX to maintain extensions, it overwrites the changes.

I would like to know where in the FreePBX code this context is generated. I might be able to modify my copy of FreePBX, if I could just find the right module and the code.

Thanks in advance for any suggestions!

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Message: 7
Date: Thu, 1 Feb 2007 08:01:41 +1000
From: ip-rob
Subject: [Amportal-devel] [new feature] assistance with incorporating
new feature
To: [email protected]
Message-ID: [email protected]

I was hoping to add a feature which would allow you to play a system recording before going to a destination. I’ve looked over the drawselects function in admin/functions.inc.php. I’ve added the code necessary to get the display using the valid list of recordings and only if the recordings module is installed.

I was going to see how to integrate this into the section where the extensions_additional.conf is written out since the playback will need to be inserted before the goto statement.

Can somone tell me where this file is generated and written out?

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Message: 8
Date: Thu, 1 Feb 2007 08:05:14 +1000
From: ip-rob
Subject: Re: [Amportal-devel] [new feature] assistance with
incorporating new feature
To: [email protected]
Message-ID: [email protected]

Forgot to add that I need to know where the records are inserted into the database as well. I’ll look, was just hoping to get a pointer or two. Thanks.

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Message: 9
Date: Thu, 1 Feb 2007 11:00:27 +1000
From: ip-rob
Subject: Re: [Amportal-devel] [new feature] assistance with
incorporating new feature
To: [email protected]
Message-ID: [email protected]

I was able to determine where the database update goes and modify the code to post the right changes to the database. I would still like to see this feature in the core system for ivr functionality only. It seems like providing a DID for people to directly reach a queue or other destination is a good message to have available in the IVR.

I decided in the short-term to create a custom destination that plays the message, then goes to the destination queue. This is a quick way to make the change I needed.

I’ll further test the code and then see if all goes well…this post is now not necessary.

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Message: 10
Date: Sun, 04 Feb 2007 13:51:57 +0100
From: Christian
Subject: [Amportal-devel] mISDN and freepbx
To: [email protected]
Message-ID: [email protected]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

hi,

i’m christian developer of chan_misdn. I’d love to see chan_misdn beeing
supported natively by freepbx. Therefore we’ve submitted a patch to the
freepbx trac (#1565).

Unfortunately until now, nobody has commented anything on this feature
request, is there anything i can do to encourage the freepbx developers
to add our patch to the current freepbx trunk?

cheers,

Christian


Message: 11
Date: Mon, 5 Feb 2007 19:42:25 +0200
From: "Mihammad Ligon"
Subject: [Amportal-devel] hi
To: "Ericka Ditzler"
Message-ID: 01c7494c$ff86b970$2101a8c0@PC340326784311
Content-Type: text/plain; charset=“us-ascii”

Hi,

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