[General] VoIP For Home Usage

[Majed B.]

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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-- 
       Majed B.