Voicemail plays even during call

odpogn

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#1
Hi all,

I recently installed Elastix with a 1FXO/0FXS x100p card from x100p.com. For the most part everything works fine, but when people call in from the PSTN line, even when I answer the call, the voicemail still plays.. Is it because I don't have a FXS port, and Asterisk can't recognize if I picked up the call?

I'm answering the calls with a regular analogue telephone that's plugged into the bypass port of the x100p.. Is that why it can't recognize a pickup? Does that mean I need to install a FXS card? or is it just some configuration issue I can easily fix?

Thanks in advance!

odpogn-
 

hinzinho

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#2
FXO port is for your telco line. FXS is for your analog phone.

It sounds like you put a Y splitter to your telco line. If so, then yes, it will not work correctly. Either get a FXS port for your analog phone or use a SIP softphone for testing.
 

blangys

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#3
You need either an IP phone or an FXS port on the system. Otherwise, the system has no way to know that you picked up the call with that bypass. The bypass is passive - it's just wired in. There is not signaling that is taking place there.

Are you using this as an answering machine for the line?
 

odpogn

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#4
Thanks for the clarification, yes we are only using it as an answering machine. It works great, until someone actually answers the phone.. then you end up hearing the voicemail during the call. Do you recommend a cheap reliable card that has both FXO and FXS to be used with an analogue telephone?
 

DaveD

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#5

dicko

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#6
odpogn said:
Thanks for the clarification, yes we are only using it as an answering machine. It works great, until someone actually answers the phone.. then you end up hearing the voicemail during the call. Do you recommend a cheap reliable card that has both FXO and FXS to be used with an analogue telephone?
Sometimes we can "over-engineer", If you are only using it as an answering machine then a cheaper and more immediate solution is to not plug the phone into the bypass port but just have both devices "bridged" on the incoming line, just set Elastix to delay answer for a few seconds it will then "just work" like a traditional answering machine.

One way to delay answer is to add:


[from-zaptel-custom]
exten => _.,1,wait(16) ; or whatever
exten => _.,n,Goto(from-zaptel,${EXTEN},1)


to /etc/asterisk/extensions_custom.conf
 

odpogn

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#7
The code just delays the answer? Then if I answer the phone, instead of playing voicemail after 10 secs, it would just play it.. say.. 30 secs later?

Or does the code stop elastix if the phone is picked up?

I'm using a splitter now and both elastix and my phone are both connected directly to the PSTN Line
 

dicko

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#8
Try it and see :) if there is no ringing on the line after 16 seconds, there is no reason to answer the ringigng that is not there.
 

odpogn

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#9
Thanks for your reply~

I tried it, but I can still hear the answering machine when I call myself from my cellphone..

HAPPY NEW YEAR~
 

dicko

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#10
Immediately or after a delay? (check the events at the CLI to make sure the wait is being executed) from-zaptel-custom is the first included line in from-zaptel, so if the calls were being answered by that context . . . if you use another context you will have to modify to suit.

Is it 2010 there yet?
 

odpogn

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#11
I think its following the general settings default delay..

The exctentions context says "from-internal" so I added:

exten => _.,1,wait(16) ; or whatever
exten => _.,n,Goto(from-zaptel,${EXTEN},1)

to [from-internal-custom] under extentions_custom.conf

I'm not able to test it out yet under [from-internal-custom] though because our phone line is being bombarded with calls right now. But I'll let you know. Before I just pasted the entire thing at the bottom of the extentions_custom.conf file.

Haha its not 2010 yet, so a little early~~
 

dicko

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#12
No "thinking" allowed (in this context it equates to guessing), knowing is much preferred :) :) , calls comes in on a trunks (not extensions), if you have setup your trunk(s) to use the "from-zaptel" context (as most who use analog lines d)o, it will use what I said as an inbound context, if you haven't then the call context will be "from-pstn" and you need to use "from-pstn-custom" context similarly.
 

odpogn

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#13
is this:

[from-zaptel-custom]
exten => _.,1,wait(16) ; or whatever
exten => _.,n,Goto(from-zaptel,${EXTEN},1)


to /etc/asterisk/extensions_custom.conf

the same as going to

PBX>PBX Configuration>General Settings>Ring Time Default = 16?

When I try to put this:

[from-zaptel-custom]
exten => _.,1,wait(16) ; or whatever
exten => _.,n,Goto(from-zaptel,${EXTEN},1)

into /etc/asterisk/extensions_custom.conf, It doesn't do anything (I'm pasting at the very bottom of the file). But if I change the Ring Default, it delays the voicemail.. but the voicemail still plays during the call.

I'm a little confused on how to check the CLI to see if its working, and what to do with this:

[from-zaptel-custom]
exten => _.,1,wait(16) ; or whatever
exten => _.,n,Goto(from-zaptel,${EXTEN},1)

The voicemail message I get is "Extention 3000 is currently unavailable, please leave a message etc etc."

If I just had an FXS port, would I be able to plug my analogue phone in and that would fix the voicemail message playing during calls?

Thank you for being patient~!
 

dicko

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#14
the wait(n) is there so Asterisk doesn't answer the call until after the delay, if the call is picked up elsewhere it won't answer, as soon as it answers it will have control over the audio part of the call as you see with the ringtimer setting.

Again, use from-zaptel-custom only if you use that as your incoming trunk context, use from-pstn-custom if you have no idea what "from-zaptel" means, thusly:

[from-pstn-custom]
exten => _.,1,wait(16) ; or whatever
exten => _.,n,Goto(from-pstn,${EXTEN},1)


instead.

The Asterisk CLI is a convenient tool for seeing the progress of your Asterisk box, you get there with

rasterisk -xvvvv

from bash. (the linux prompt) and watch it is you "do stuff" with the system, it will write out each parsed line from your dialplan as the logic is followed.

http://voip-info.org

has much fuller desrcriptions and recipes.
 

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