[General] VoIP For Home Usage

[Ahmad Al-Ibrahim]

Majed,

Yes, that was the first thing to do, next I configured the PBX to be on DMZ
to have all of the incoming traffic.

Regards,

Ahmad Al-Ibrahim

On Fri, 31 Jul 2009 13:53:25 +0300, "Majed B." <majedb at gmail.com> wrote:
> 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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