[General] VoIP For Home Usage

Tue Aug 4 03:10:26 +03 2009

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

Majed B. wrote:
> Ahmad,
> 
> How did you add the scripts? Can you provide the details please?
> 
> I got mine to work with my friend in the US using the settings I used
> for NATting in my previous emails.
> Instead of using an IP, I'm using a sub domain under my domain name
> which automatically gets updated by my firewall (IPcop). My dynamic
> DNS domain name is handled by no-ip.com.
> 
> Keep us updated on OpenVox's usage: Installing, configuring, sound
> quality, ...etc.
> 
> Good luck!
> 
> On Sun, Aug 2, 2009 at 8:39 PM, Ahmad Al-Ibrahim<ahmad at koutbo6.com> wrote:
>> I got it to work finally.
>> with
>> nat=yes
>> externip=<router WAN IP>
>> localnet=192.168.1.0/24
>> and added scripts to update externip in /etc/asterisk/sip_nat.conf value
>> every 5 minutes and reload sip without killing active calls using
>> asterisk -r -x "sip reload"
>> from shell
>>
>> I'm gonna install OpenVox card tomorrow inshalla!
>>
>> Thanks for your help guys.
>>
>> Regards,
>>
>> Ahmad Al-Ibrahim
>>
>> Ahmad Al-Ibrahim wrote:
>>> I've assigned single port to RTP without luck.
>>>
>>> Regards,
>>>
>>> Ahmad Al-Ibrahim
>>>
>>> On Fri, 31 Jul 2009 07:36:13 -0700 (PDT), Faisal AlAbhoul
>>> <kuwait at q8net.com>
>>> wrote:
>>>> Like Bashar said Zain wont give you a real IP, instead they bridge your
>>>> device with the real ip at their end, i have found that while trying to
>>> fix
>>>> my brother's router to host online games.
>>>>
>>>> Ahmad, try to allocate a single port to RTP instead of a range, this
>>> might
>>>> help as i have done it on my ClarkConnect router.
>>>>
>>>> --- On Fri, 7/31/09, Bashar Al-Abdulhadi <bashar at kuwaitnet.net> wrote:
>>>>
>>>> From: Bashar Al-Abdulhadi <bashar at kuwaitnet.net>
>>>> Subject: Re: [General] VoIP For Home Usage
>>>> To: "General OpenSource Discussion" <general at oskw.org>
>>>> Date: Friday, July 31, 2009, 5:19 PM
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Zain NAT's your mobile connection maybe thats causing the issue ?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Ahmad Al-Ibrahim wrote, On 07/31/2009 01:12 PM:
>>>>
>>>>   Majed, I went through most of the articles, my case is th last and the
>>>> worst case, where asterisk is behind NAT, and the mobile is behind NAT
>>>> in other network.
>>>>
>>>> tcpdump shows RTP packets sent from asterisk to the mobile without a
>>>> response from the mobile.
>>>>
>>>> Regards,
>>>>
>>>> Ahmad Al-Ibrahim
>>>>
>>>> Majed B. wrote:
>>>>
>>>>
>>>>     Ahmad, the issue has already been addressed before and is documented
>>>> with solutions. The first 3 hits should have the proper solution for
>>>> you/us:
>>>>
>>>> http://www.google.com/search?q=asterisk+nat
>>>>
>>>> Let me know which one works out!
>>>>
>>>> I just installed IPcop on my HP T5700 ThinClient (which has support
>>>> for my SpeedTouch USB DSL Modem)!!! So I should be able to get
>>>> Asterisk running online in 2 days or so. Inshallah....
>>>>
>>>> On Fri, Jul 31, 2009 at 3:58 AM, Ahmad Al-Ibrahim<ahmad at koutbo6.com>
>>> wrote:
>>>>
>>>>       Majed,
>>>>
>>>> Yes, that was the first thing I've tried without success, after that I
>>>> configured the PBX to be on DMZ. I will let you know once it is solved.
>>>>
>>>> Regards,
>>>>
>>>> Ahmad Al-Ibrahim
>>>>
>>>> Majed B. wrote:
>>>>
>>>>
>>>>         Ahmad: Did set port-forwarding rules from your gateway to
>>> Asterisk?
>>>> According to what I read, these ports should be redirected to the
>>>> Asterisk box:
>>>>
>>>> 5060 (tcp/udp)
>>>> 10000-20000 (udp)
>>>>
>>>> On Fri, Jul 31, 2009 at 3:31 AM, Faisal AlAbhoul<kuwait at q8net.com> wrote:
>>>>
>>>>
>>>>           Thats the drawbacks of SIP 2.0 over NAT. Try to use IAX2
>>>>           protocol, its very
>>>> NAT-Friendly but hardly any software clients for mobile phones support
>>> IAX
>>>> or IAX2.
>>>>
>>>> I see you didnt install G729, you can download the opensource G729 and
>>>> simply place it in modules directory. The G711 is actually the alaw and
>>>> ulaw
>>>> codec, perfect for LAN telephony and FAX over VoIP.
>>>>
>>>>
>>>>
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>>>>
>>>>
>>>>
>>>>
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>>>>
>>>>
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> 