search for: shibuya

...Is there anybody else who has build Asterisk on Opteron? Please share your experiences with the community. This is important! You should also submit your changes to Digium in order to get them into the CVS tree. Please contact Mark about that. thx rgds benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __________________________________________________ GANBARE! NIPPON! Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE http://mail.ganbare-nippon.yahoo.co.jp/

...nk you again. However, there hasn't been a single response from a graphic designer to offer help with a custom icon. Are there any graphic designers on this list at all? If so, please take a look at the Wiki above and see if you can help. thanks rgds benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __________________________________________________ GANBARE! NIPPON! Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE http://mail.ganbare-nippon.yahoo.co.jp/

Does anybody out there have any evidence that Sipura support is still in operation? rgds 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.

...another installer package for the new version. Also, I am still working on extending the install package so that users can choose whether or not they want to install the sources. Anybody interested in this, please bare with me a few more days. regards benjamin -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __________________________________________________ Do You Yahoo!? http://bb.yahoo.co.jp/

...ful. This will only happen if Solaris users start sharing their stuff in a place where others can easily find it. So, please share your experiences with the community ... http://www.voip-info.org/tiki-index.php?page=Asterisk+Solaris+Support thanks rgds benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __________________________________________________ GANBARE! NIPPON! Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE http://mail.ganbare-nippon.yahoo.co.jp/

...another installer package for the new version. Also, I am still working on extending the install package so that users can choose whether or not they want to install the sources. Anybody interested in this, please bare with me a few more days. regards benjamin -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __________________________________________________ Do You Yahoo!? http://bb.yahoo.co.jp/ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/ast...

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc.

...> vpb/1-4: Event [12=>[03] Loop Drop] vpb/1-4: Flushing event [12]=>[03] Loop Drop > vpb/1-4: handle_notowned: mode=3, event[12][[03] Loop Drop ]=[0] > vpb/1-4: handle_notowned: mode=3, [12=>0] thanks in advance regards benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __________________________________________________ GANBARE! NIPPON! Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE http://mail.ganbare-nippon.yahoo.co.jp/

...and Zaptel, but the false hangups persist. BTW, the customer does not wish to convert his ISDN line to analog lines for a variety of reasons. Does anybody know how to adjust the zaptel driver so that false hangups can be reduced? thanks in advance regards 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.

Is there a way to override the SIP From Header that is used in the extension.conf Dial command? The default is 'asterisk@host'. I do not want to configure SIP accounts in sip.conf, but instead generate the SIP From-User within extensions.conf from data the user has entered interactively. Any idea? Henrik -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ody know how many meters of cable is supported between > > Asterisk Digium-FXO cards and the analog telephone ? > > You don't plug analog phones into FXO cards, that's not what FXO cards > are for. > > rgds > 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. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digiu...

Hi there, I have an X-Lite phone on my box and I'm trying to register it with a remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I know it's a pain to do because of SIP not working well with NATs, but I know there are ways to do such a thing...moving the Asterisk box outside the NAT is not a possibility at the moment. One thing we tried was setting up a VPN, but

Hello, I'm researching the possibility of using VOIP (SIP) with an existing PABX system. Ideally, the setup would be to dial an outside line through the PABX (that would actually link to the the VOIP gateway). At this point I would prefer not to purchase a hardware-based VOIP gateway. I would prefer to use a software-based gateway for research & testing purposes. Could anyone please

...t need this patch. More detail can be found in the ReadMe section of the installer. The patch is available for download at http://www.astmasters.net/stuff/AsteriskMissingFilesPatch.pkg.tgz My apologies for any inconvenience this may have caused. rgds benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __________________________________________________ Do You Yahoo!? GANBARE! NIPPON! http://mail.ganbare-nippon.yahoo.co.jp/

...now, we don't wish them ill, but Vonage seems to have been out of commission for quite a little while. Website is excruciatingly slow, log-in fails, hard-line and soft-line are out (alert-tone or "subscriber not in service") -- even "Network failover" is failing. Outgoing calls return fast-busy. 866-support number is busy. Let's see how they deal with that in

...g ease of use for flexibility in order to get a complete newbie started. This is not a guru tool. Just in case you wondered, this will be released under the GPL and is intended to become part of the Asterisk distribution. thanks for your interest regards benjk -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __________________________________________________ GANBARE! NIPPON! Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE http://mail.ganbare-nippon.yahoo.co.jp/