[General] VoIP For Home Usage

[Faisal AlAbhoul]

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
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