[pp.int.general] Using voice server for PPI

[Gregory Engels]

Greetings all,

The Pirate Party of Hesse have an Asterisk voice server
running, that can be used for international communication.
the 8xxx rooms on the conferece server are reserved for this
very task.

The german wiki page for it is here: http://wiki.piratenpartei.de/HE:Telefonkonferenz

I need to translate the wiki page (well the non technical part of it)
to english.

The server also allows for a telephoneconference for being streamed
and recorded in the free OGG format.

Here the actual dial-Ins:
PSTN Germany +49 69 175 36 743 (Sipgate)
PSTN Switzerland: +41 31 5110514 (PhoneStar*)
PSTN Austria: +43 720 512348 (Sipcall)
PSTN Luxembourg: +352 203 330 004 638 (MIXvoip)
you can also use a SIP-softphone to dial

  sip:500 at sip.piratenpartei-hessen.de (German prompts)
sip:501 at sip.piratenpartei-hessen.de (English prompts)
sip:502 at sip.piratenpartei-hessen.de (French prompts)
sip:503 at sip.piratenpartei-hessen.de (Spanish  prompts)
sip:504 at sip.piratenpartei-hessen.de (Italian prompts)
sip:505 at sip.piratenpartei-hessen.de (Swedish prompts)
sip:506 at sip.piratenpartei-hessen.de (Russian prompts)

or you can use SipBroker: http://www.sipbroker.com/sipbroker/action/pstnNumbers
after a prompt dial *9846501 (or instead of 501 for english any other  
language out of the list above)
But in my experience there had been difficulties to reach some of its  
numbers.

Since we would use the room 8000# for our board meetings, we could
have a stream at http://sip.piratenpartei-hessen.de:8000/raum8000.ogg

(I put somewhat more work in this email than needed, since i can use the
text for the wiki page straight away. )

So, if you have a SIP Provider we should add, I would be glad to  
arrange this.
It should be as follows:

If there is a SIP provider that is providing a FREE PSTN dial in  
number (also known
as DID = Direct Inward Dialing) AND it is allowed for someone who is not
from that country to subscribe AND it is not limited for just personal  
use,
AND it should allow more than one simultaneous phone call,
then write me a link back to the regstration form and i work my way  
through it.

If any of the conditions above could not met, you are free to  
subscribe yourself
and submit the username, password and sip-server login to me and we  
will add
it then to the voice server.

If you already have an asterisk server, we could set up a connection
between the servers (this would need more time for testing to ensure
perfect interoperability and stability)

If there is no free SIP provider in your country, then you can either
use a sip- softphone. A list can be found here:
http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software#SIPClientsUAs

it may be needed with some soft-phones to enter a sip subscription,
in that case you can get one from one of the providers that we are
using for our server, like  http://www.PhoneStar.ch

If you know of an free asterisk language package that is good,
write to me and we will add a language on the server. (the problem
is, there are sometimes low quality packages out there, and we
cannot distinct say a joke package from high quality one, if we
do not speak that language)

If you think you can benefit from it, you can try it anytime. If you
need a moderated access, please submit a request for a
dedicated room, stating the purpose and who should get
admin password to me, and it will be granted on the next
board meeting of the pp-Hesse. (usually every second Wednesday)



greetings all,


Gregory Engels
International Coordinator
Pirate Party of Germany

mobile: +49 172 853 44 91
skype:gregory.engels
jabber:dichter at jabber.piratenpartei.de





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