[General] VoIP For Home Usage

[Faisal AlAbhoul]

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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