[General] VoIP For Home Usage

[Ahmad Al-Ibrahim]

You weren't able to register? or make calls?

Try capturing packets to see whats wrong, maybe the ISP blocking port 5060?

Regards,

Ahmad Al-Ibrahim

Majed B. wrote:
> Ahmad,
> 
> How did you add the scripts? Can you provide the details please?
> 
> I got mine to work with my friend in the US using the settings I used
> for NATting in my previous emails.
> Instead of using an IP, I'm using a sub domain under my domain name
> which automatically gets updated by my firewall (IPcop). My dynamic
> DNS domain name is handled by no-ip.com.
> 
> Keep us updated on OpenVox's usage: Installing, configuring, sound
> quality, ...etc.
> 
> Good luck!
> 
> On Sun, Aug 2, 2009 at 8:39 PM, Ahmad Al-Ibrahim<ahmad at koutbo6.com> wrote:
>> I got it to work finally.
>> with
>> nat=yes
>> externip=<router WAN IP>
>> localnet=192.168.1.0/24
>> and added scripts to update externip in /etc/asterisk/sip_nat.conf value
>> every 5 minutes and reload sip without killing active calls using
>> asterisk -r -x "sip reload"
>> from shell
>>
>> I'm gonna install OpenVox card tomorrow inshalla!
>>
>> Thanks for your help guys.
>>
>> Regards,
>>
>> Ahmad Al-Ibrahim
>>
>> Ahmad Al-Ibrahim wrote:
>>> I've assigned single port to RTP without luck.
>>>
>>> Regards,
>>>
>>> Ahmad Al-Ibrahim
>>>
>>> On Fri, 31 Jul 2009 07:36:13 -0700 (PDT), Faisal AlAbhoul
>>> <kuwait at q8net.com>
>>> wrote:
>>>> Like Bashar said Zain wont give you a real IP, instead they bridge your
>>>> device with the real ip at their end, i have found that while trying to
>>> fix
>>>> my brother's router to host online games.
>>>>
>>>> Ahmad, try to allocate a single port to RTP instead of a range, this
>>> might
>>>> help as i have done it on my ClarkConnect router.
>>>>
>>>> --- On Fri, 7/31/09, Bashar Al-Abdulhadi <bashar at kuwaitnet.net> wrote:
>>>>
>>>> From: Bashar Al-Abdulhadi <bashar at kuwaitnet.net>
>>>> Subject: Re: [General] VoIP For Home Usage
>>>> To: "General OpenSource Discussion" <general at oskw.org>
>>>> Date: Friday, July 31, 2009, 5:19 PM
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Zain NAT's your mobile connection maybe thats causing the issue ?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Ahmad Al-Ibrahim wrote, On 07/31/2009 01:12 PM:
>>>>
>>>>   Majed, I went through most of the articles, my case is th last and the
>>>> worst case, where asterisk is behind NAT, and the mobile is behind NAT
>>>> in other network.
>>>>
>>>> tcpdump shows RTP packets sent from asterisk to the mobile without a
>>>> response from the mobile.
>>>>
>>>> Regards,
>>>>
>>>> Ahmad Al-Ibrahim
>>>>
>>>> Majed B. wrote:
>>>>
>>>>
>>>>     Ahmad, the issue has already been addressed before and is documented
>>>> with solutions. The first 3 hits should have the proper solution for
>>>> you/us:
>>>>
>>>> http://www.google.com/search?q=asterisk+nat
>>>>
>>>> Let me know which one works out!
>>>>
>>>> I just installed IPcop on my HP T5700 ThinClient (which has support
>>>> for my SpeedTouch USB DSL Modem)!!! So I should be able to get
>>>> Asterisk running online in 2 days or so. Inshallah....
>>>>
>>>> On Fri, Jul 31, 2009 at 3:58 AM, Ahmad Al-Ibrahim<ahmad at koutbo6.com>
>>> wrote:
>>>>
>>>>       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> 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
>>>> General at oskw.org
>>>> http://oskw.org/mailman/listinfo/general_oskw.org
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>   _______________________________________________
>>>> General mailing list
>>>> General at oskw.org
>>>> http://oskw.org/mailman/listinfo/general_oskw.org
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> -----Inline Attachment Follows-----
>>>>
>>>> _______________________________________________
>>>> General mailing list
>>>> General at oskw.org
>>>> http://oskw.org/mailman/listinfo/general_oskw.org
>>> _______________________________________________
>>> General mailing list
>>> General at oskw.org
>>> http://oskw.org/mailman/listinfo/general_oskw.org
>>>
>> _______________________________________________
>> General mailing list
>> General at oskw.org
>> http://oskw.org/mailman/listinfo/general_oskw.org
>>
> 
> 
>