[General] VoIP For Home Usage

[Ahmad Al-Ibrahim]

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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