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Asterisk zaptel pulse dialing

Created by: gonzo,Last modification on Wed 29 of Mar, 2006 [11:38 UTC] by sybren

Pulse Dialing on Zap Channels

Since 1.0.1 Asterisk has pulse dialing support for zapata channels. Just specify

pulsedial=yes

in zapata.conf


Workaround for Incorrect Pulse Decoding

If you are using FXS ports on a TDM400 card and are having problems with Asterisk not recognizing pulses, or misinterpreting the pulses that are dialed, you may need to perform the following steps (as of Zaptel v1.2.0):

1) Open the file wctdm.c from the Zaptel source in a text editor.

2) Locate the lines that read:

/* Reset the debounce (must be multiple of 4ms) */
wc->mod[card].fxs.debounce = 8 * (4 * 8);

(These can be found in the neighborhood of line 950.)

4) Then, cut this value (the pulse debounce time limit) in half by changing the wc-> line to the following:

wc->mod[card].fxs.debounce = 4 * (4 * 8);


5) Compile and install the Zaptel package and make sure that the new wctdm module is loaded. Asterisk should now receive pulse dials accurately and reliably.

(This solution was provided to me by Max on the asterisk-users list and has made my Asterisk 1.2.0 machine correctly receive pulse-dialed digits every single time.)


(Outgoing) Pulse Rates

Although most telephone exchanges are quite happy with rates of anywhere from 6 to 15 pulses per second (pps), the standard accepted by telephone companies is only 8 to 10 pps. Some modern digital phone exchanges, free of the mechanical inertia problems of older phone systems, will accept a pps rate as high as 20.

Besides the pps rate, the phone dialing pulses have a make/break ratio, usually described as a percentage, but sometimes as a straight ratio. The North American standard is 60/40 percent; most of Europe accepts a standard of 63/37 percent. This is the pulse measured at the phone, not at the phone exchange, where it's somewhat different, having traveled through the phone line with its distributed resistance, capacitance, and inductance. In practice, the make/break ratio does not seem to affect the performance of the phone dial when attached to a normal loop.

There are three parameters to adjust pulse dialing make/break ratio when Asterisk sends pulse digits over an FXO interface.
So, configuration for European telephone lines will look like:

zaptel.pulse.make: 60
zaptel.pulse.break: 40
zaptel.pulse.pause: 800


The FreeBSD driver may allow you to adjust these values using sysctl.


See Also


Other Asterisk development by PortaOne


Asterisk

Comments

Comments Filter
222

333Pause in pulse dial fix

by litnimax, Monday 29 of December, 2008 [05:13:35 UTC]
See http://bugs.digium.com/view.php?id=13999
222

333Patch is ok but ...

by , Wednesday 03 of November, 2004 [12:40:09 UTC]
I tried patch with Asterisk 1.0.2 and Zaptel 1.0.2 and it is ok. Pacth utility reported that it successfully installed all the diffs. However, when I try to dial out, it doesnt break the dial tone which is typical issue when you have tone dialing device dialing pulse line (in case of regular phone you just change the settings on the phone set from tone to pulse). When I do zap show channel I see that it still reports that channel is not pulse based, despite the fact that I changed my zapata.conf with pulse=yes. It looks like this is not solved with the patch at all.
222

333patch final release pb !

by , Tuesday 26 of October, 2004 [20:21:12 UTC]
me too !! Dose anybody success with 1.0 final ?
222

333How to patch final release 1.0?

by , Saturday 25 of September, 2004 [11:25:07 UTC]
I am unable to patch new release 1.0 nor 1.0 RC1 and 2. Dose anybody success and how?