similar to: [SPAM] - RE: Grandstream-Budge tone - Email found in subject

Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli

You'll need one of the Digium Cards to make/receive pstn call...Chose the best one fits to you... To make calls over ip, voip phone is also needed such as grandstream, cisco, etc. or you can download x-lite from xten.com And use it with your computer as a soft phone... _____ From: infra struct [mailto:infrastructt@yahoo.com] Sent: Monday, June 06, 2005 1:01 PM To:

Hi; Can I use two TDM04B in one asterisk box with asterisk@home? Thank You Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi m?mk?n olmayabilir. Mesaji alan kisi, mesajin g?nderilmek istendigi kisi veya kurulus

Hi; Have two grandstream Budge tone...Connected them to the network and able to make call to/from them. But when the coming call answered, I can not hear any voice and also my voice is not heart... I am able to hear voice only if I pressed the hold button and take the call again....This problem also Occurs in calls from x-lite to cisco7940... Does anybody has any idea or documentation

Hi all of you. I am trying to configure voice mail in asterisk and i am facing problems. I have found following warning message in /var/log/asterisk/messages -------------- No application 'Voicemail' for extension (macro-mainmenu, s, 5) I have configured voice mail accordingly in extention.conf [headoffice] -- ------------ ------------- exten => _63,1,Macro(mainmenu)

hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call

I have a Cisco 7940G IP Phone. I am trying to load the firmware to SIP 3.2. The Phone just hangs in Defaulting CM to TFTP Server. It doesn't do anything else after that. I also have two other 7940g's that are the Universal Application Loader mode and say Protocol Application Invalid. I need to know what I can do to fix both these problems. I am running Asterisk@Home version 1.0. I have to

Hi, I am using ajax_scaffold to generate against a model backed by atable on SQLServer... It creates all the stuff, and I can list the items in my table... But, the "Edit" and "Create New" buttons dont work... The strange thing is that the problem occurs when I edit the ajax_scaffold generated controller with before_filter/set_charset... I try to set the charset of pages

Hi, I am new to Rails. And I wonder how to define lookup fields for a table... Suppose we have an employee table and a departmant table... Each employee record have a department_id field... And when listing employees, we want to show the user friendly department_name from department table.. How can I do this? Thanks .... Bu e-posta mesaji, mesajin alici kisminda belirtilmis

Hi, I use scaffold to generate against a model and controller.. Then, I modify the controller code, and rewrite paginate method with :select => ["col1, col2, col5"] so that it only retrieves the columns I wanted... But, :select does not work and it seems that it selects all the columns of the underlying table... What am I missing ? Thanks... Bu e-posta mesaji,

Hi, I am using ajax scaffold generator with charset=iso-8859-9(Turkish)... Edit & Create buttons work correctly when there is no turkish chars in the current record. But, if there are turkish chars in any of the fields in the current row, buttons don''t work... (loading indicator simply spinning forever...) I put the KCODE=''u'' line in

I can't setup Authentication between samba & Win NT [root@fulya etc]# net join -U Administrator%pass [2003/08/12 20:22:55, 1] utils/net_ads.c:ads_startup(176) ads_connect: Connection refused [2003/08/12 20:22:56, 1] utils/net_rpc.c:run_rpc_command(154) rpc command function failed! (NT_STATUS_ACCESS_DENIED) Joined domain TRANSVARO. [root@fulya root]# What is ads_.. ? if it is

Hi, We have Ubuntu 14.04.3 server and Samba 4.1.6-Ubuntu. Windows and Mac Os X clients connect this server. Some Os X users sometimes report "error code -43" when file copying to server. What can cause the problem? this is my smb.conf [global] workgroup = hurguc realm = HURGUC.LOCAL preferred master = No security = ads password server = dbdns2.hurguc.local encrypt passwords =

HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files . What are the configurations has to be made with asterisk ? Thanx in advance, Luke. Send instant messages

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

I don't off-hand remember, but the folks on the user list probably do (;-)) You accidentally sent to the development list --dave Serhat G?RSOY wrote: > *********************** > This mail and its attachments have been scanned by InterScan MSS. > ***********-*********** > > > hi folks, > > I am a new samba user and I have a problem like this; > > When I copy

I have no clue what this package is for, but reading the help page for the ccomp function tells you that it returns a numeric vector or matrix. How do YOU want to display information from this numeric vector? That will determine how you would put it into a data frame. -- Sent from my phone. Please excuse my brevity. On November 22, 2017 3:13:01 AM PST, Levent TERLEMEZ via R-help <r-help at

Hi, I'm using Mandrake 10 CE, Samba 3.02a, Winbind. I have got a strange problem. My linux machines can not join to our NT 4.0 domain when I define a netbios name which includes "I" or "i". How can I solve this problem. Any answers .... Thanks a lot. Domain Name = DOMAIN1 Netbios Name = LINUX1

Dear Users, I would like to use compositions package with ggplot/ggtern, other composition classes of compositional package can be used with ggtern by converting to data frame but I could do anything with c(ount)comp class. Ggplot/ggtern can not recognise comp and also can not be converted to data frame. Is there any other way to do this? Thank you in advance, Levent TERLEMEZ.

Dear Users, What is the proper way to write symbol, superscript, subscript in ggtern/ggplot? I tried every given example, every possible features of ggplot but couldn?t achived. I just want to write P_a, sigma^2, etc, would you please advise me about this problem. Thanks in advance, Levent TERLEMEZ ________________________________ Bu elektronik posta ve onunla iletilen b?t?n dosyalar sadece