Incoming call routing

[tawny]

Really don't get this! Can someone please clear this up for me:

At present, only two phones ring on incoming trunk calls, and then go to the generic 'press x for all phones in house' etc.

I want to:

1) Make ALL phones ring
- I have looked at the default inbound route in FreePBX (custom app, with a string pointing to somewhere in pluto). Can't find any reference to this in Linux MCE. Have got a group that 'rings all', 600, but can't seem to point the inbound route to this as it won't let me??!

2) Either change the IVR or remove it.
- I've found the call routing section of LinuxMCE but cannot understand it as it refers to people rather than trunks etc. Maybe I'm missing something but...? Also, here I have tried setting '# of seconds to ring before ivr' to zero and also 99 in an attempt to disable it but nothing! Why am I feeling stupid right now?

[maverick0815]

From what you write, I guess you have turned off automatic configuration. I have been running around that area myself before.
If you want all your phones to ring, you need to turn on automatic configuration and use the linuxmce interface...there is no real use going to freepbx for that.
With automatic config on, you will see your configured phoneline and there are some links right next to it...click on settings and you can tick all the phones that you wish to ring.
If I remember correctly only the orbiterphones are ticked there by default. After you have done that, click on telecom->user call routing and make the appropriate settings for your users..which phones will ring for what users etc.
Other than that, I would not fiddle with Freepbx unless you know exactly what you do. I only went there to modify my sip-trunk...all the other stuff is supposed to be handled by linuxmce itself

[tschak909]

Do not modify the IVR response timeout, as it is currently an integral part of the default dialplan context. If you set it to 0, it will fall through completely to hang up.

I'm growing very irritated at people with some asterisk knowledge immediately fucking shit up to suit their view of the universe, yet they won't take that same time that they've spent hours upon hours upon end trying to work with the team to try and support more configuration, instead expecting us to support it out of the box.

I'm tired of it.

-Thom

[pw44]

Hi,
if IVR response timeout shall not be modified, IMHO, it shouldn't even exist.....
But, again, IMHO, the default timeout is too short, making it almost impossible to pick up the phone before IVR assumes if you are not close to the phone, which is a PITA.
I do live in Brazil, where portuguese is spoken, lmce IVR is in english, most people don't understand what is happening and simply hang up.
Some hints in regard of how to change the default language of IVR and have a translated voice would be very much appreciated.
Well, in such a case, i would like to have a suggestion of what is to be done, in order to let the phone rings at least for 40 seconds before IVR assumes.
Also please, could you inform the default values for the IVR timeout (webamin AND freepbx admin)?
Regards,
Paulo

[tschak909]

You can increase it, you just can't set it to 0.

-Thom

As for prompts in other languages, are you $(#@@ serious in that you _EXPECT_ us to be able to support it?

Everybody has their pet features that HAVE to be supported...

Why don't you get off your ass and help us add support for other languages in the prompts?

[pw44]

No, i'm not asking you the be able to support it...
I'm just asking what is to be done (lmce team created voice files, if it exists) and where, so i can implement it, and if it works, i will gladly make it available for all, not only in portuguese, but also in german (i do speak both).
I can help, but i'm far to be a freepbx specialist, that's why i'm asking, before doing it myself and risking $(#@@ my installation.
So, if you can inform if there are voice files created by pluto / lmce team, and where they are located, i can help.

[tschak909]

Check out a copy of the source tree from here:

http://svn.linuxmce.org/svn/branches/LinuxMCE-0810/src/

and go into src/Asterisk

There is a perl script to generate the voice menu, in that folder. Modify, copy to /usr/pluto/bin to test.

When you're ready, you can run svn diff against it, make a ticket, and submit the diff as a patch.

-Thom

[pw44]

generate_users_voicemenu.pl ?

[tschak909]

...

Yes. Look inside it.

-Thom

[pw44]

Is there any parameter which defines the language?
I did find PK_country, but did not find where the defined country for the core is stored.
Multiple countries speaks the same language.. may i suggest any modification or extension?

[tschak909]

what would you propose?

-Thom

[pw44]

Like having field for country spoken language, for telephony purposes, i.e....
Germany, Austria and Switzerland (part) speaks german, Brazil speaks brazilian portuguese, Portugal, Angola, Mozanbique, Timor speaks portuguese of Portugal, Spain, Central and South America (excluding Brazil) speaks spanish.....
This way, the generate_users_voicemenu.pl could get the language definition from the database and get the language library from lmce....
I'm doing some changes on this script, will test it during the weekend and after done, will diff it.
The languages i'm adding for pluto initial IVR are german and brazilian portuguese, and for the asterisk voices, i already got the full library for german and brazilian portuguese, which could (i think) be selected by an asterisk config parameter?

[tawny]

Thanks - have enabled automatic config again, and can now select all extensions. However, I can only see the sipgate line; not the other two lines I have configured. I guess I need to recreate the trunks outside of FreePBX so that LMCE is aware of them. I've read this wiki article so can follow this, but don't see how this would give me an option to add a second line - am I missing something? Sorry if my actions are adding to your annoyance Thom, but as someone who has been installing PBX and VoIP equipment for well over a decade, I'd like to think that I'm 'knowledgeably adjusting settings via a commonly used GUI' rather than 'fucking shit up'. Everybody has different requirements, I know.

The IVR announcement is still playing after just a few rings, despite me having changed the answer time to 99. Any ideas? No offence Thom, but as an Englishman speaking 'original English', I can't understand a single word of what the IVR announcement is saying. It's dreadful. I guess this could be changed but would lose the text to speech functionality which I imagine is the reason it sounds synthesised. Could I do better? I don't know, but I'd rather just not use it anyway.

[tschak909]

There are a few choices here:

(1) change the voice used. We provide all of the current festival voice sets.
(2) fix the dialplan to allow for IVR to be disabled, submit patch.
(3) add code to allow for custom IVR to be recorded, submit patch.

-Thom

[los93sol]

I've been pretty deep on the telecom stuff recently, and what you have to realize, it that there are still parts of the telecom system using voices we don't even generate currently.  At this time, the best option is a combination of Thom's suggestions to allow the IVR to be disabled completely AND to allow for custom recordings.  The key here is that you have to do all of the translations which includes the sos security menus and would seemingly include the voicemail menu's as well to provide a unified user experience.  I'm not really in favor of even starting down this road until we have stabilized and unified the current voice menu's.  That said, if you're going to get down and dirty, stop by the dev channel and partner with us to get this stuff fixed, but remember, we need to fix everything currently in the telecom system before we start adding more complexity to it.