similar to: ISDN in Japan?

Hi All, I currently run asterisk in our office (in Japan) and use a cisco PRI gateway for connection to the PSTN. I would like to setup some more systems for our smaller offices (in Japan) that would use BRI and preferably using a PCI card in the asterisk box and not a seperate Cisco gateway (expensive). HOWEVER, Japan has this INS64 protocol for their BRI lines and im not sure what cards

Hi, R experts. I am a new user of R and trying to learn this program. I have a problem. Here is the code. d<-as.Date(c("2000/01/03","2000/01/05","2000/01/19","2000/01/28")) r<-rnorm(4) da<-data.frame(d,r) a<-as.Date("01/01/2000","%d/%m/%Y") b<-as.Date("30/01/2000","%d/%m/%Y") ab<-seq(a,b,by=1)

Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --------------------------- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP

Hi everyone! I'm wondering if anyone on the list had the opportunity to work with an NTT INS1500 ISDN PRI service before. You see, in Japan, if you receive a call that was just forwarded by another number, the call presentation not only includes the caller (ANI) and your number (DNIS), it will also usually include the forwarding number (REDIRECT). Does anybody know how to extract this field

Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Anybody here having trouble receiving email from the list on Gmail? I havn't received anything since Friday July 29.

Hello - I am looking at connecting 7 - 10 locations together using Asterisk and possibly some VoIP gateway appliances. I need to insure best voice quality as these trunks will be used primarily for customer calls. I am considering implementing a full T1 frame relay circuit to each location which can be done for a reasonable cost. DSL and Cable are currently at each location and setup for

Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/311c56ec/attachment.htm

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Sebastian and all, This is a Japanese to live in Japan. I have a question. Ogg's window functon is f(x)=sin(1/2*pi*sin^2((x+0.5)/n*pi)) for 0 <= x < n But I cannot understand why this window function satisfys the equation f^2(x)+f^2(x+n/2)=2 which is for MDCT's perfect signal reconstruction. reference: Analysis/Synthesis Filter Bank Design Based on Time Domain Aliasing

We have written some monitoring and stat collection scripts that use asterisk -rx "command" The script runs once a min and logs data and posts any critical notifications. Everything is working well with this method but we get the -- Remote UNIX connection / disconnect message once a min and we would like to suppress it. Is it possible without reducing the verbose logging level.

Hello there, I tried to found documentation about openloop disconnect on Asterisk/Zaptel. And up to now, I didn't find anything. Is openloop disconnect supported by zaptel/wcfxo drivers? Other question: maxsilence and silencethreshold don't seem to have any effect on voicemail. Could it be possible that in certain situation it just wont work? I tried adding some verbose in

Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? Do they support things like WPA etc?? I have heard the battery life can be a problem.. Is this the case? Thanks..

Can anyone recommend any reading to help me get a better understand of VOIP technology and all the terms used in this discussion thread? I've read many documents on the wiki but I find it a bit lacking. Sincerely, Tyler