similar to: VOIP in Japan using Freebit

Hi, My setup is: IPPhone1 --- Asterisk1 with B410P ---- Patton 4638 --- Asterisk2 --- IPPhone2 I want to evaluate Asterisk1 in TE/PtmP mode. So, Patton box is configured in NT/PtmP (with 3 BRI links between both systems). Anyway, asterisk -rx "pri show spans" keeps replying : PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active PRI span 3/0: Provisioned,

Hi thanks to everybody who has been assisting me in solving the various problems I had to dial out from Asterisk to a PSTN number with SIP using Nikotel's VoIP service. I have drafted a mini-how-to which is available at http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf This is a first draft, I will amend this further, in particular the "verify and debug" section

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk

Hi Sunil, I had filed a bug and saw your response stating that it would be fixed in version 14. In the meanwhile , what we want to know is whether this bug is a minor bug and can be ignored for now. Does reporting 100% inodes cause any problem for the OCFS file system or can we ignore this bug and go into production. Also can you tell us by when version 14 would be released. R'gds

Hi! > How to do this ?? > To proceed with your answer on PICKUPMARK, where do I put this ??? Look at the example for Asterisk 1.4 on this page: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup Philipp

Currently f_fsid of struct kstatfs returned from ocfs2_statfs() is undefined (vfs layer fills 0 as default). Since in some conditions, f_fsid value might be used as (f_fsid, ino) pair to uniquely identify a file, ocfs2 should return a unique defined f_fsid value from ocfs2_statfs(). Because uuid_str is identified no mater on big or litlle endian machine, it's also endian consistent to use

Currently f_fsid of struct kstatfs returned from ocfs2_statfs() is undefined (at least it should be filled with 0). Since in some conditions, f_fsid value might be used as (f_fsid, ino) pare to uniquely identify a file, ocfs2 should return a defined unique f_fsid value from ocfs2_statfs(). This patch uses uuid_hash as a unique ID to initiate f_fsid value, the 32bits width is enough for ocfs2

I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to "connected" even the callee is still ring. How could asterisk distinguish the "ringing" and "connected" in zap channel? thanks.

...and everybody is struggling with the globalization/localization. I''ve been monitoring the Japanese rails mailing list and there''s a Tokyo study group already drawing attention. I think they''re having their fourth meeting soon. There''s a couple toy applications out there and I believe cookpad.com is currently being rewritten with rails in the background.

Dear Community member, (Sorry for the spam earlier, triggered a send by keyboard shortcut) the Xen Project was planning to have a booth at LinuxCon Japan this year (for more information, see

Hi All, We will hold a Linux IO controller Mini-Summit in Tokyo, Japan on Saturday, Oct. 17, 2009 -- day before the 9th Linux kernel summit. This is a good opportunity to meet face-to-face to discuss issues and directions for developing IO controller. Details of the summit are still sketchy, so suggestions for agenda topics are welcome. Please e-mail to me or post on the wiki page at

Hi All, We will hold a Linux IO controller Mini-Summit in Tokyo, Japan on Saturday, Oct. 17, 2009 -- day before the 9th Linux kernel summit. This is a good opportunity to meet face-to-face to discuss issues and directions for developing IO controller. Details of the summit are still sketchy, so suggestions for agenda topics are welcome. Please e-mail to me or post on the wiki page at

Hi All, We will hold a Linux IO controller Mini-Summit in Tokyo, Japan on Saturday, Oct. 17, 2009 -- day before the 9th Linux kernel summit. This is a good opportunity to meet face-to-face to discuss issues and directions for developing IO controller. Details of the summit are still sketchy, so suggestions for agenda topics are welcome. Please e-mail to me or post on the wiki page at

Hi, Kerry Garrison from Fonality will be with us live to address the Trixbox so-called "phone home" script issue. I'm going to try to have something about the year "2007 in review" for any and all VOIP and Asterisk-related events, so if anyone wants to report on what they've been doing in 2007, you're welcome to chime in. We're also talking about doing a

Hi all, I hope there is someone in Japan. If we install Centos in Japanese, and then I ssh to it from an English client. Will the SSH prompt be in Japanese? How to make the SSH prompt in Japanese? Thank you. -- Fajar.

Hello, I?m working with an employer that is looking to hire someone to fulfil a permanent DevOps-type position at their office in Tokyo. Japanese language is not required; only English. Consequently I had hoped that some members of this mailing list may like to discuss further; off-list. I can be reached using "JamesBTobin (at) Gmail (dot) Com". Kind regards, James

Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100505/5068aaab/attachment.htm

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Ok, latest Wine Version (0.9.25), new OS (Ubuntu Daper Drake (6.06)) Here is my console output: //--------------------------------------------------------------------------- fixme:ole:OaBuildVersion Version value not known yet. Please investigate it ! fixme:ole:CoRegisterMessageFilter message filter has been registered, but will not be used fixme:ole:OleLoadPictureEx