[General] VoIP For Home Usage

[Ahmad Al-Ibrahim]

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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