[General] VoIP For Home Usage
Majed B.
majedb at gmail.com
Fri Aug 7 13:45:10 +03 2009
Alright. I got PBX in a Flash to finally dial out of the landline just
now. Seems like all I was missing is including the config file that
"genzaptelconf" generated.
If you get a message when trying to dial out of the landline: "All
circuits are busy now"
Then login to the shell and type: asterisk -rvvvvvvvvvv (more v gets
more verbose)
Type: zap show channels
You should see something like this:
pbx*CLI> zap show channels
Chan Extension Context Language MOH Interpret
pseudo from-internal en default
1 from-zaptel en default
2 from-internal en default
3 from-internal en default
4 from-internal en default
If you get the pseudo channel only, then it means your configuration
isn't included in zapata.conf
Simply type: nano /etc/zapata.conf
Scroll down till you see this:
;Include genzaptelconf configs
#include zapata-channels.conf
Go above the semicolon line (comment) and add this:
;Include genzaptelconf configs
#include zapata-auto.conf
Save the file (CTRL+X, Y to agree, Enter to save) and run: amportal restart
This will reload asterisk with the new settings.
Login back to asterisk (asterisk -rvvvvvvvvvv) and type: zap show channels
If all is good, quit and dial out. It should work now.
After seeing this, I think I have a chance with Asterisk 1.6 again.
I'll give it another quick shot and get back to you.
* Note: I added two variables to zapata.conf
busydetect=yes
busycount=5
This will detect a busy tone and automatically hangup.
You can find more settings here:
http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf
I found the solution to the inclusion of the config file here:
http://www.trixbox.org/forums/trixbox-forums/open-discussion/2-3-0-10-problem-when-installing-not-automatically-detecting-m
On Fri, Aug 7, 2009 at 3:42 AM, Majed B.<majedb at gmail.com> wrote:
> On my tests today I saw that trixbox 2.8 does have support for OpenVox
> cards. So what you have is just a misconfiguration somewhere.
> (Hopefully).
>
> On Fri, Aug 7, 2009 at 3:39 AM, Ahmad Al-Ibrahim<ahmad at koutbo6.com> wrote:
>> Nope, Just 1.6, I'm planning to try 1.4.
>>
>> Majed B. wrote:
>>> Have you tried using Asterisk 1.4? Or just 1.6?
>>>
>>> On Fri, Aug 7, 2009 at 3:32 AM, Majed B.<majedb at gmail.com> wrote:
>>>> I have no clue at all about the values of these, but since the caller
>>>> ID settings of UK worked here, try using those.
>>>>
>>>> You WILL get static if you don't attach a micro-filter (splitter) to
>>>> the phone when attaching it to the FXO port, when the DSL modem is
>>>> hooked. So, always use the micro-filter.
>>>>
>>>> On Fri, Aug 7, 2009 at 3:21 AM, Ahmad Al-Ibrahim<ahmad at koutbo6.com> wrote:
>>>>> I will try that, thanks for the tips.
>>>>>
>>>>> I cannot say kw settings did not work with asterisk 1.6 since the
>>>>> problem might be with the line and the card.
>>>>>
>>>>> If I connect the line directly to the FXO module I always get busy tone.
>>>>> If I use ADSL splitter, I get continuous Starting/Hangup in asterisk
>>>>> CLI, and the line rings, and the call is answered, all I hear is phone
>>>>> pickup, hangup every second.
>>>>>
>>>>> One thing I'm not sure of is the DMTF and MF settings, if you look at
>>>>> the attached zonedata.c file in previous emails, last 4 variables in
>>>>> Kuwait zone are:
>>>>> .dtmf_high_level = -10,
>>>>> .dtmf_low_level = -10,
>>>>> .mfr1_level = -10,
>>>>> .mfr2_level = -8,
>>>>>
>>>>> These values were a guess. Do you have any idea on the values of these
>>>>> variables for Kuwait?
>>>>>
>>>>> Regards,
>>>>>
>>>>> Ahmad Al-Ibrahim
>>>>>
>>>>> Faisal AlAbhoul wrote:
>>>>>> Some cards need 4-wires to be powered, by joining 4-wires (2-wires on
>>>>>> each end) from the wall socket you'll end up powering the 4 lines in the
>>>>>> RJ11 socket in your Vox card, i remember this being done in a forum and
>>>>>> worked for some cards. its like flipping a coin but worth trying if all
>>>>>> you need to do just change your wiring.
>>>>>>
>>>>>> So the "kw" setting did not work with Asterisk 1.6 i take it?
>>>>>>
>>>>>>
>>>>>> --- On *Thu, 6/8/09, Majed B. /<majedb at gmail.com>/* wrote:
>>>>>>
>>>>>>
>>>>>> From: Majed B. <majedb at gmail.com>
>>>>>> Subject: Re: [General] VoIP For Home Usage
>>>>>> To: "General OpenSource Discussion" <general at oskw.org>
>>>>>> Date: Thursday, 6 August, 2009, 10:34 PM
>>>>>>
>>>>>> 2 gray cables mean you get 4 lines. (1 line for every 2 wires).
>>>>>>
>>>>>> If you connect an analog phone, do you get a tone? If not, you need to
>>>>>> reverse the wires, otherwise your setup is fine.
>>>>>>
>>>>>> I'm not sure about what Faisal suggested though for the 4 wires. All
>>>>>> the RJ11 cables I have have 4 pins, but only 2 wires are connected,
>>>>>> and it did work before with Asterisk 1.4.
>>>>>>
>>>>>> On Thu, Aug 6, 2009 at 10:28 PM, Ahmad Al-Ibrahim<ahmad at koutbo6.com
>>>>>> </mc/compose?to=ahmad at koutbo6.com>> wrote:
>>>>>> > Interesting, I checked the wall socket, there are 2 gray wires each of
>>>>>> > them having 4 colored wires, and only 2 colored wires of each gray
>>>>>> wires
>>>>>> > are connected to the socket (2 red wires to the right, 2 blue wires to
>>>>>> > the left) 4 other wires are not connected. 2 in each gray wires.
>>>>>> I'm not
>>>>>> > familiar with wiring. Now do I have to connect the others? IIRC the
>>>>>> > others are not used/utilized.
>>>>>> >
>>>>>> > Regards,
>>>>>> >
>>>>>> > Ahmad Al-Ibrahim
>>>>>> --
>>>>>> Majed B.
>>>>>>
>>>>>> _______________________________________________
>>>>>> General mailing list
>>>>>> General at oskw.org </mc/compose?to=General at oskw.org>
>>>>>> http://oskw.org/mailman/listinfo/general_oskw.org
>>>>>>
>>>>>>
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>>>>>>
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>>>>>
>>>>
>>>>
>>>> --
>>>> Majed B.
>>>>
>>>
>>>
>>>
>>
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>
>
>
> --
> Majed B.
>
--
Majed B.
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