Kuniyoshi Murata
2004-Sep-14 05:48 UTC
[Asterisk-Users] Use ISP's SIP account for IP-PSTN gateway
Hi, I'm thinking of introducing Asterisk on Linux for IP PBX. Now I'm using ISP that has VoIP service and I have VoIP terminal box for that ISP and a SIP account for SIP server of the ISP. Now, what I would like to do is the following. A. Setup IP PBX on Linux by using Asterisk. B. For IP-PSTN gateway, configure Asterisk to use my ISP's SIP account and connect to my ISP's IP telephony service. Is that possible? -- Kuniyoshi Murata.........................iChat/AIM:macwebcaster English-Japanese Interpreter mailto:kuni@ej-interpreter.net Macintosh Webcast Specialist http://www.macwebcaster.com
Greg Hill
2004-Sep-14 05:52 UTC
[Asterisk-Users] Use ISP's SIP account for IP-PSTN gateway
On Tue, 14 Sep 2004, Kuniyoshi Murata wrote:> I'm thinking of introducing Asterisk on Linux for IP PBX. > > Now I'm using ISP that has VoIP service and I have VoIP terminal box for > that ISP and a SIP account for SIP server of the ISP. > > Now, what I would like to do is the following. > > A. Setup IP PBX on Linux by using Asterisk. > B. For IP-PSTN gateway, configure Asterisk to use my ISP's SIP account and > connect to my ISP's IP telephony service. > > Is that possible?Hey, this is Linux. It can do anything. If you can code it. :-) You'd just have to set up a context for that service in your sip.conf. Probably need to register=> with the service as well. You'll need your sip username and password in order to do this. Look at the examples in the sample sip.conf, or look through the mailing list archive for example sip.conf's. Greg
Benjamin on Asterisk Mailing Lists
2004-Sep-14 06:20 UTC
[Asterisk-Users] Use ISP's SIP account for IP-PSTN gateway
On Tue, 14 Sep 2004 21:48:52 +0900, Kuniyoshi Murata <kuni@ej-interpreter.net> wrote:> A. Setup IP PBX on Linux by using Asterisk. > B. For IP-PSTN gateway, configure Asterisk to use my ISP's SIP account and > connect to my ISP's IP telephony service. > > Is that possible?Murata-san, A is certainly possible, B depends on how you want to connect and what provider you are using. You can get a Digium X100P FXO card and connect it to the RJ-11 analog phone jack on your ADSL modem. Digium's card does not have JATE type approval but if you use an ADSL splitter so that you are not connecting through to NTT, then you don't need any JATE approved card. Otherwise, you need to get a winmodem from a Japanese importer with JATE type approval, but it must have the Intel 537 chipset (aka Ambient) which is very hard to get these days in Japan. Kuroutoshikou used to sell a modem with this chipset but they stopped selling them. In principle you could use Asterisk's ability to talk to SIP proxies and connect to your SIP provider directly without any analog converter boxes in between, but in practise this is "muzukashii". The providers don't easily give you the necessary information and even if they do, it usually doesn't work. At present, the only VoIP service in Japan that doesn't expect you to use an analog telephone adapter (inside your ADSL modem) is Taraba.net but they are at present only accepting connections from X-Lite and X-Pro softphones. This is expected to change in the near future though as they are getting their customer support ready to deal with other types of devices. If you have any more questions about Asterisk or need any assistance feel free to contact me directly. I am in Shibuya. Nihongo demo daijoubu desu. yoroshiku benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.