[General] VoIP For Home Usage
Ahmad Al-Ibrahim
ahmad at koutbo6.com
Sun Aug 2 23:22:02 +03 2009
Majed,
Try using your WAN IP in the externip and see how it goes. I made a call
with a friend in Argentina for 30 minutes. Voice is crystal clear!
Use the following python script to update your sip_nat.conf, I suggest
backup up the target file, the script is going to write on it.
You need to download PyDNS package from
http://pydns.sourceforge.net/
I have this script running every 5 mins.
--- begin ---
#!/usr/bin/python
import os
import time
import DNS
hostname='your_host_name'
server='208.67.222.222' # using OpenDNS to resolve
localnet='192.168.1.0/24' # your LAN
file_path='/etc/asterisk/sip_nat.conf'
try:
request = DNS.Request(name=hostname, server=server, qtype='A')
response = request.req()
ip_address = response.answers[0]['data']
f = open(file_path, 'w')
f.write('nat=yes\n')
f.write('externip=%s\n' % ip_address)
f.write('localnet=%s\n' % localnet)
f.close()
time.sleep(2)
# soft sip reload without dropping calls
os.system('asterisk -r -x "sip reload"')
except:
pass
--- end ---
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.
>>>>
>>>>
>>>>
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