[General] VoIP For Home Usage

[Ahmad Al-Ibrahim]

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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