Portuguese Asterisk Users Group Blog Share and learn experiencies about VoIP telephony and Asterisk PBX.

Wednesday, July 05, 2006

Mark Spencer at Astricon Berlin

Asterisk

Stefan Wintermeyer interviewed Mark Spencer (the inventor of the famous Asterisk PBX) at an Asterisk convention in Berlin.

How old are you and when did you start to use Linux and becoming a part of the open-source community?
I am currently 29 years old and got involved with Linux in 1994 at the age of 17.

When, why and how did you start the Asterisk project?
I started Asterisk in 1999 for the purpose of being our internal PBX for "Linux Support Services", now called Digium. It wasn't until 2001 that the company changed its name to Digium and refocused around Asterisk.

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Tuesday, July 04, 2006

New project of Loop test Spans


Hi All.
Below the new project of Loop test, between spans, remembering that one span must be configured with signalling to master with signalling= pri_net and to another one must have the signalling= pri_cpe, functioned perfectly without no error of signalling, for 48 hours.

Bolts of the Loop handle version 2
Bolts A---------------------Bolts B
1 4
2 5
4 1
5 2

Best Regards

Josué

Thursday, June 29, 2006

Cabo para testar Link E1

Pessoal, saudações a todos.
Esse post somente orienta aqueles que possuem uma TE205P/TE210P ou mesmo uma TE405P/TE410P, para auxiliar no teste entre spans.
É possivel ser testado link´s E1 ISDN PRI e mfc/r2 com este cabo.
Pinagem Ponta A Ponta B
1 - B. Verde Azul
2 - Verde B. Azul
3 - B. Laranja B. Laranja
4 - Azul B. Verde
5 - B. Azul Verde
6 - Laranja Laranja
7 - B. Marrom B. Marrom
8 - Marrom Marrom

Saudações a todos, se precisarem de algo que eu possa ajudar é só comunicar.

Josué

Sunday, June 25, 2006

Asterisk handling My Skype Calls

Call me!

Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html .

Main features that any one can easily integrate into Asterisk:

- Route skype incoming calls into Asterisk DialPlan, then you just can do ANYThing route to your Meetme rooms, IVRs do it in your way.

- Dialout calls from any SIP extension through Skype (reaching Skype contacts or outgoing calls to landline through Skype Outgoing calls prices.

- Enable your website with SkypeMe Button and route it to Asterisk! Feel free to listen MusicOnHold from my Asterisk Box through my Skype Account.

Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype.


MoutaPT