[General] VoIP For Home Usage

Wed Aug 5 01:56:57 +03 2009

Ahmad, here's what I noticed about the NAT thing.

It's NOT a NAT thing. It's a DNS resolution problem. Whenever I run
"dig domain.net" on the PBX machine, the PBX recognizes the public IP
of my domain for a few minutes, then it goes back to unknown.

So instead of having script update the IP for me, I'll just run dig
every 3-5 minutes... It's retarded but it works, until I figure out
what the hell is going on with this.

Even the guys at #asterisk & #freepbx aren't helping on this...

Which reminds me, you could go there and ask them. They're on
irc.freenode.net, port 6667. Register your nickname to access their
channels:

/nick <your nickname>
/nickserv register <pass> <email>

Login to your email and copy/paste the verification line

/nickserv identify <pass>

On Tue, Aug 4, 2009 at 7:44 PM, Majed B.<majedb at gmail.com> wrote:
> I don't see where the problem is Ahmad. The manual says to use wires 4
> & 5, which are what we use in Kuwait (The middle 2 wires) and it works
> on my end.
>
> Notice that they have shown a figure of 8 wires not 4; that's because
> they have digital services, unlike us.
>
> With that said, Kuwait's caller ID is similar to that of UK (BT
> company), so the way that someone showed in the asterisk forum might
> actually work for you:
>
> 1          ------1
> 2---------/
> 3---------\
> 4          ------4
>
> http://www.asterisk.org/forum/viewtopic.php?p=42449&sid=f7f76ad03a8a80004579d26d75ae1206
>
> But before you venture into making such a cable, open the phone's wall
> socket and look at where the cables are wired (which pins). You should
> follow the same.
>
> If you confirm the wiring, then focus on the PBX itself. I haven't
> found anything that points at where the problem is, apart from these
> two links:
> http://www.mail-archive.com/asterisk-gui@lists.digium.com/msg01430.html
> http://www.mail-archive.com/asterisk-gui@lists.digium.com/msg01394.html
>
> They say that if you had old trunk configurations, and deleted them,
> the GUI may not have cleaned them up as it should (notice the date of
> these messages: They're in July 2009).
>
> So you have two options, as suggested in the links:
> 1) Delete the trunk configuration then dive into the specified config
> files and make sure there are no references of them.
> 2) Reinstall.
>
> Good luck!
>
> On Tue, Aug 4, 2009 at 4:45 PM, Ahmad Al-Ibrahim<ahmad at koutbo6.com> wrote:
>> Majed,
>>
>> This is the installation guide link
>> http://www.openvox.com.cn/downloadsFile/1156846942.pdf
>> The manual talking about RJ45 (UK setup), which is not the case for me,
>> the card has 4 RJ11 ports.
>>
>> Card model is A400P
>>
>> Regards,
>>
>> Ahmad Al-Ibrahim
>>
>> Majed B. wrote:
>>> I have my country settings set to default: United States of America. I
>>> did try switching it to UK once to see if that would make caller ID
>>> work, but it didn't so I put it back to USA.
>>>
>>> As far as I understand, all lines in the world ring the same, but they
>>> send data differently. 24volts in the loop & 90volts on ringing.
>>>
>>> I think you should read the manual of the card you bought. It may
>>> require some wire-swapping, as the Asterisk forum link suggests. Does
>>> it say anywhere on the box of OpenVox whether it's using US or UK
>>> version of making the adapter?!
>>>
>>> What's your card's exact model number?
>>>
>>> On Tue, Aug 4, 2009 at 4:34 AM, Ahmad Al-Ibrahim<ahmad at koutbo6.com> wrote:
>>>> Some posts suggest using a modem cable rather than phone cable.
>>>> http://forum.voxilla.com/linksys-sipura-voip-support-forum/spa3000-no-dial-tone-15732.html#post81622
>>>> http://www.asterisk.org/forum/viewtopic.php?p=42449&sid=f7f76ad03a8a80004579d26d75ae1206
>>>> not sure if this apply to us.
>>>>
>>>> Regards,
>>>>
>>>> Ahmad Al-Ibrahim
>>>>
>>>> Ahmad Al-Ibrahim wrote:
>>>>> I have configured the trunks, inbound routes, zap channels. It seems to
>>>>> be something related to the line it self or the configurations to match
>>>>> Kuwait PSTN tones, are you able to call the land line? and do you see
>>>>> debug information on asterisk CLI while calling the land line? What is
>>>>> the 'country' variable value in indications.conf, 'loadzone' and
>>>>> 'defaultzone' variables values in zaptel.conf?
>>>>>
>>>>> Having a cronjob wont harm your system, atleast it works.
>>>>>
>>>>> I've already saw this link in the morning, I dont have acpi variable set
>>>>> in grub.conf.
>>>>>
>>>>> Regards,
>>>>>
>>>>> Ahmad Al-Ibrahim
>>>>>
>>>>> Majed B. wrote:
>>>>>> I was just about to email and ask you how things are going on your end! hahaha.
>>>>>>
>>>>>> Well, seems like I wasn't the only one going through trouble...
>>>>>>
>>>>>> Have you configured the trunk, made sure the Outbound route is using
>>>>>> that trunk, and that you have configured a general Inbound route? Did
>>>>>> you configure an extension to receive/make calls?
>>>>>>
>>>>>> I know it sounds silly, but go over them again and try recreating the
>>>>>> extension or make a new one and divert the inbound calls to it.
>>>>>>
>>>>>> I'll look around and see what could cause your problems.
>>>>>>
>>>>>> As for my side, remember that NAT & externip thing? It worked for a
>>>>>> while, when asterisk wanted it to, then it broke. If in the home page
>>>>>> the value "Public IP" is "Unknown" then you can kiss your audio good
>>>>>> bye on any device outside the LAN from receiving audio. I'm too close
>>>>>> to stab someone in the face... it's quite frustrating because my
>>>>>> settings should work, and I don't want to run a script with a cron job
>>>>>> to replace the domain name with the IP :/
>>>>>>
>>>>>> On Tue, Aug 4, 2009 at 3:10 AM, Ahmad Al-Ibrahim<ahmad at koutbo6.com> wrote:
>>>>>>> Majed,
>>>>>>>
>>>>>>> I've installed OpenVox card, ran setup-pstn, the card was detected and
>>>>>>> configured, but I'm having problems, not sure what else needs to be done.
>>>>>>>
>>>>>>> I keep getting lots of
>>>>>>>
>>>>>>>    -- Hungup 'DAHDI/1-1'
>>>>>>>    -- Starting simple switch on 'DAHDI/1-1'
>>>>>>>    -- Hungup 'DAHDI/1-1'
>>>>>>>    -- Starting simple switch on 'DAHDI/1-1'
>>>>>>>    -- Hungup 'DAHDI/1-1'
>>>>>>>    -- Starting simple switch on 'DAHDI/1-1'
>>>>>>>
>>>>>>>
>>>>>>> messages in asterisk console when the line is connected behind an ADSL
>>>>>>> splitter, and when i call the land line number the phone rings, pick up
>>>>>>> and hangups every second simultaneously with the messages above. I've
>>>>>>> removed the ADSL splitter and connected the line directly to the box, I
>>>>>>> keep getting busy tone.
>>>>>>>
>>>>>>> Do I have to configure the device specifically for Kuwait PSTN tones?
>>>>>>>
>>>>>>> BTW Dahdi is zaptel in Asterisk 1.6
>>>>>>>
>>>>>>> Regards,
>>>>>>>
>>>>>>> Ahmad Al-Ibrahim
>>>>> _______________________________________________
>>>>> General mailing list
>>>>> General at oskw.org
>>>>> http://oskw.org/mailman/listinfo/general_oskw.org
>>>>>
>>>> _______________________________________________
>>>> General mailing list
>>>> General at oskw.org
>>>> http://oskw.org/mailman/listinfo/general_oskw.org
>>>>
>>>
>>>
>>>
>>
>> _______________________________________________
>> General mailing list
>> General at oskw.org
>> http://oskw.org/mailman/listinfo/general_oskw.org
>>
>
>
>
> --
>       Majed B.
>

-- 
       Majed B.