How to setup Elastix 2.0 inbound route with 1 FXO?

leiw3248

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#1
Hi,

I tried to add phone number in DID field, but it don't work.

Please help !
 

fmvillares

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#2
dont understand the question so its impossible to help with so little explain.
 

leiw3248

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#3
Hi,

I remember 1.6.0 version set phone number to DID field and set phone number to ZAP channel, the inbound call only goto this channel. But 2.0 version don't work ...

Please help !
 

fmvillares

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#4
yes it works..unembedded freepbx...
 

leiw3248

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#5
How to set it in unembedded freepbx?
 

fmvillares

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#6
the same as before
 

leiw3248

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#7
I just tested if don't enter phone number in DID field, it work. But enter phone number it dosen't work.
 

fmvillares

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#8
you have configured dahdi trunks or dahid zap compatibility trunks
 

leiw3248

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#9
I don't need two Elastix connected, why have to use dahdi trunks.

I don't need analog phone, why have to use dahid zap compatibility trunks.
 

ramoncio

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#10
leiw3248 said:
Hi,

I remember 1.6.0 version set phone number to DID field and set phone number to ZAP channel, the inbound call only goto this channel. But 2.0 version don't work ...

Please help !

Hi Leiw3248, maybe you are confused or confusing, I don't know. There you talked about assign did to zap channel, this is for fxos, hence analog lines.

Could you please explain a little better and deeper your environment and problems?

Also, please have a read here:

http://www.catb.org/~esr/faqs/smart-questions.html
 

leiw3248

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#11
Hi,

I'm testing to separate two phone number with two FXO, one for voice and one for fax, so they can't effect each others. How to setup them?

Thanks !
 

fmvillares

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#12
what? if they are 2 phone numbers they are 2 lines..u dont need to separate them as they are separated just now
 

leiw3248

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#13
Hi,

How to setup inbound route and outbound route with following:

1. Phone01 dial out with line01 (can't random to line02), also external user call line01 will ring phone01?

Thanks !
 

fmvillares

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#14
the best way to begin is to erad elastix without tears...there u will find the correct answer as it is perfectly explained
 

leiw3248

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#15
Hi,

I read it already and tried to setup it, but still fail, please help !
 

faraheneh

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#16
hi,
can you give more details so we can help better? what are your configs and what are the system errors?
I have followed elastix without tears steps and there was no problem so if you give more details, the answers will be more accurate...

regards,
 

leiw3248

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#17
Hi,

Could you give me step by step on Elastix 2.0 ?

Thanks !
 

faraheneh

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#18
hi,
I wrote a really really detailed answer for you but when I pressed the submit button an error occurred and they were deleted :(
I don't have much time now but soon I will post them again, so before that I wanted to confirm your scenario, what I guessed was this:
you have to FXO modules and you want to have one of them for send/receive fax and another to make outbound calls. you have a phone01 that has an extension -like SIP- and it is registered in elastix and works fine and you want all the inbound routes be redirected to phone01. -you don't need and IVR-
your elastix server has detected your modules and you have no problem with them.
was I right? and by the way I checked elastix 2.0, the configs are mostly like elastix 1.6... and the step by step manual of elastix without tears is really good... which part is giving you problem?
 

leiw3248

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#19
Hi,

I don't understand how to setup inbound call to phone01, for example: line01 to phone01 and line02 to phone02, how to separate them ? if not separate them, will Elastix use default inbound route to randomly route inbound call to two phones?

Thanks !
 

dicko

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#20
leiw3248:

In the replies to your 410 posts, you have frequently been referred to "Elastix Without Tears", I can but ask you "have you actually read it yet?", I can but suspect that your answer would have to be a resounding "No" because it works for 96.2% of everyone else who cared to go that route, The other 3.8% forgot to take their Ritalin.

It is totally laid out there for you, very specifically and with examples. So you need to put your analog trunks in the "from-zaptel" context, you need to use FreePBX to assign appropriate pseudo DID's to the channels you have, these numbers should probably be your phone numbers that match your FXO's this is only for your convenience , you can call them 1234 and 1235 if you want. You then need to use the standard "inbound route" module to match those pseudo DID numbers that YOU defined to where YOU want them to go to, if YOU want the pseudo zap-tel DID that YOU programmed as 1234 to ring YOUR extension 4567 then so program it!! When a call comes in on one of your FXO's (dahdi/n where n is the enumeration of your telepheno lines on the hardware ports) it will so be directed there. Which part of the above are YOU still having problems with?

With respect, maybe you needs to drop a few bucks on the PaloSanto "paid support" link below as you are stillm after all this time apparently completely out of your depth and haven't reasonably taken advantage of any of the free advise you have recieved. Do you not agree leiw3248?

My guess is it will take them all of five minutes for them to "fix you up".
 

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