[General] VoIP For Home Usage

Majed B. majedb at gmail.com
Fri Jul 31 23:54:26 +03 2009 

I'm behind my thinclient which is acting as a firewall (NATting).

I saw that link and the current hostname is: pbx.localdomain
There's no mechanism for me to put my domain name and have the IP
auto-change, unless I write a shell script to do that periodically.

Also to note: When you change the hostname in the trixbox GUI from the
System menu, it doesn't necessarily change it in the hosts file (!!).
So make sure you always change it from the web interface and not the
command line, as it seem to be written and used in other places.

I forgot to ask my friend if he's behind a nat or not.

On Fri, Jul 31, 2009 at 11:48 PM, Ahmad Al-Ibrahim<ahmad at koutbo6.com> wrote:
>
> Try setting the hostname of the pbx in /etc/hosts
>
> check this out
> http://www.trixbox.org/forums/trixbox-forums/help/nat-problems-changed-sipnatconf-and-still-no-audio
>
> are you behind NAT?
> your friend is using a public IP or behind NAT?
>
> Regards,
>
> Ahmad Al-Ibrahim
>
> On Fri, 31 Jul 2009 23:14:22 +0300, "Majed B." <majedb at gmail.com> wrote:
>> I was just able to call my friend in the US (I created him an
>> extension on my PBX after a few settings)...
>>
>> I can hear him clearly but my voice is very faint to him.
>>
>> Also, I'm unable to dial out anymore. Even to my voicemail and the
>> stupid clock announcer....
>>
>> I added these lines in the file: /etc/asterisk/sip_general_custom.conf
>>
>> nat=yes
>> externip=pbx.mbhbox.net
>> localnet=192.168.0.0/255.255.255.0
>>
>> On Fri, Jul 31, 2009 at 10:56 PM, Ahmad Al-Ibrahim<ahmad at koutbo6.com>
>> wrote:
>>>
>>> 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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>>>
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-- 
       Majed B.

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