Problems with Outbound PSTN calls

mikedmr

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#1
Hi,

I've experienced line drops on occasional but not always, outbound calls made through the FXO ports, when i call PSTN lines.

I'm using the latest Elastix Distro and a 4 port digium compatible card.

Incoming calls from PSTN lines are fine.

Line drops in a sense that calls are cut short. Is there a time limit here i should know about?


Thanks
 

danardf

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#2
It's strange your problem! :huh:

Try to increase the rxgain into your dahdi config file.
 

dicko

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#3
There is no "time limit" in Asterisk by default, (you can however define one).

Analog telephony has been around for way more than a century. The concept is little changed.

trouble shooting is relatively easy , a "butt" set, a volt/ohm meter, and understanding of FXO/FXS

# FXO (Foreign Exchange Office)
A voice interface, emulating a PABX extension, as isit appears to the C.O. (Central Office) for connecting a PABX extension to a multiplexer.
# FXS (Foreign Exchange Subscriber)
A voice interface, emulating the extension interface of a PABX (or subscriber interface of a C.O.) for connecting a regular telephone set to a multiplexer.
(blatantly stolen from somewhere)

Basically if your regular analog phone doesn't exhibit that behavior but your Elastix box does, then I suggest:
a) Ensure that any disconnect supervision you are using is the same as your provisioner's.
b) If you use busydetect, then make sure busycount is not too low (gotta hate forklifts)
(both these settings are in the zapata/dahdi conf files)

If these are all fine then suspect your "hardware", as essentially Asterisk only speaks "primitives" (the zaptel/dahdi driver which sends this signalling to the hardware, generally these drivers are quite mature and isn't known to cause such problems).

"digium compatibility" is necessary for sangoma, rhino, or any vendor's hardware to use these drivers, unfortunately the physical interface to the phone line is outside any part of these drivers, so you are then at your manufacturer's mercy/competence/QC.


So I guess I'm suggesting that before you spend your time with Asterisk debugging (there is some high level processed debug info, but it is only interpretive of what the hardware is telling it).
Ensure that all voltages, currents and impedances are within the norms of your country as to FXO and FXO ports/signalling at the two-wire (telephone pair) level first.

As a hint a regular analog phone bridged on that phone-pair can often be more diagnostic than many more sophisticated methods (and likely to substantiated danardf's (Hi Franck, you beat me to it!) need for TX/RX gain settings.
 

danardf

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#4
Yes, Hi Dicko. How are you?

It's true that the phone on each country are different.
for exemple, in France, the specificity are: -48V, 15mA, and the phone must have 2,2µF with 600ohms. The ring candence under 75~RMS : 1,5" / 3,5". And it's the phone that regulates the line to 15mA.
 

dicko

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#5
Franck, Ju suis tres bon, et tu (mamon et bebe aussi) ?

We agree on the basics but when it comes to line impedance current-loop, de-emphasis and indications alors!, mais vive la difference!

Also I remind you ATT comes before ITU/ETNS both alphabetically and numerically. :)


JFTHOI:

http://www.epanorama.net/circuits/telei ... ml#general
 

danardf

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#6
dicko said:
Franck, Ju suis tres bon, et tu (mamon et bebe aussi) ?
J'aime bien ton Français, C'est charment. :p
(I'm not better than you in English)
How are you = Comment allez vous? So answer:
"Je vais bien" and not "Ju suis tres bon" because you say "I'm very good" :p
Correct: Ju suis tres bon => Je suis très bon => Je vais bien.
Et tu => et toi
Mamon => la maman
et bebe => et bébé

Yes, my wife and baby are very well. The baby is always hungry. :laugh:


We agree on the basics but when it comes to line impedance current-loop, de-emphasis and indications alors!, mais vive la difference!

Also I remind you ATT comes before ITU/ETNS both alphabetically and numerically
Ho... i forgot:
The tone is 440Hz
 

dicko

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#7
Merci, c'est un long temps depuis l'ecole.

and we have different "ringer equivalency", we even have a few "ground starts" left, we call them "party lines" :laugh:
 

danardf

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#8
Merci, c'est un long temps depuis l'ecole.
But, congratulations on your effort. :)
It's very well.
 

mikedmr

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#9
Thanks.

I will try the RXgain thing. Do you think outbound calls to the pstn will also be cut short if call waiting is enabled on that particular pstn phone line?

mike
 

danardf

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#10
I don't know, but why not.
On FXO line you don't have the call waiting!
If you have one call, you can't make another call on the same line.
 

mikedmr

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#11
Hi again,

I tried the RX gain thing, but it did not fix the problem. Right now I have both inbound and outbound PSTN calls being cut short randomly.
I will post my settings here. Maybe you guys can point me to the right direction.

Thanks!
 

dicko

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#12
Did you ever check the proper functioning of your analog lines? (in the absence of Elastix)

If you did and they are clean (no hum clicks or buzzes) and of the right level (voltage and volume) (i.e. a regular analog phone works invariably and correctly)

then from the asterisk cli

core set debug 99

and explore your /var/log/asterisk/full file after an "event" , there will be lots there, but buried in there will be a reason for the disconnect on the dahdi/zap channel.
 

fraggle4

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#13
You may be experiencing "talkoff". Check that relaxdtmf=yes is not set in zapata.conf (or chan_dahdi.conf
 

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