[General] VoIP For Home Usage

[Majed B.]

Ahmad,

Did you port-forward UDP: 5060,10000-20000 to your Asterisk box?

As for the sound, you need to change the codec.

On Fri, Jul 31, 2009 at 1:26 PM, Ahmad Al-Ibrahim<ahmad at koutbo6.com> wrote:
> Bo Rashed,
>
> I've tried that without luck, I'm trying to connect to asterisk from my
> mobile using 3G/GPRS which is behind NAT, and asterisk is behind NAT too.
>
> trixbox1*CLI> sip show peers
> Name/username              Host            Dyn Nat ACL Port     Status
> 300                        (Unspecified)    D   N   A  5060     UNKNOWN
> 200/200                    192.168.1.8      D   N   A  5060     OK (74 ms)
> 100/100                    95.66.116.121    D   N   A  5060     OK (946 ms)
>
> The device is registered with asterisk without problems, calling will
> ring both ways, the problem is with the sound, RTP not being connected
> properly.
>
> Regards,
>
> Ahmad Al-Ibrahim
>
> Faisal AlAbhoul wrote:
>> Try adding the following two lines in your sip_nat.conf  in /etc/asterisk/
>>
>> externip= <your router ip address>
>> localnet=x.x.x.0/255.255.255.0
>>
>> This can solve many problems especially if you are trying to receive
>> inbound calls through the internet.
>>
>>
>> --- On *Fri, 7/31/09, Ahmad Al-Ibrahim /<ahmad at koutbo6.com>/* wrote:
>>
>>
>>     From: Ahmad Al-Ibrahim <ahmad at koutbo6.com>
>>     Subject: Re: [General] VoIP For Home Usage
>>     To: "General OpenSource Discussion" <general at oskw.org>
>>     Date: Friday, July 31, 2009, 3:58 AM
>>
>>     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
>>     </mc/compose?to=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.