[General] VoIP For Home Usage

[Ahmad Al-Ibrahim]

Majed

I have already done what is there in the page. no luck.

I use USR9108 DSL modem, it supports dyndns, and can update the record for
you :)

Regards,

Ahmad Al-Ibrahim

On Fri, 31 Jul 2009 21:12:38 +0300, "Majed B." <majedb at gmail.com> wrote:
> 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
>>
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