[General] VoIP For Home Usage

[Majed B.]

Ahmad,

Check out this page:
http://www.trixbox.org/forums/trixbox-forums/open-discussion/sip-nat-conf-and-externip

Apparently, what I had in mind actually works! This will save me the
trouble of updating the IP every time the DSL modem disconnects (if it
did), and since IPcop has the ability to integrate with dyndns and
no-ip, it will update my host by itself. Ain't that sweet? :D

On Fri, Jul 31, 2009 at 5:46 PM, Faisal AlAbhoul<kuwait at q8net.com> wrote:
> I noticed you're having 74 ms within your LAN, if its the mobile device i
> think its normal but otherwise its unusual. The extension 100 having 900ms,
> that's an odd case indeed....was Asterisk just restarted or Trixbox  heavy
> downloading when you got the debugging page?
>
> --- 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, 1:26 PM
>
> 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.
>>
>>     _______________________________________________
>>     General mailing list
>>   &nb
>
> _______________________________________________
> General mailing list
> General at oskw.org
> http://oskw.org/mailman/listinfo/general_oskw.org
>
>



-- 
       Majed B.