similar to: Can log into the mailbox from Soft-phone , but not from Hardware Phone

Hi I am using a 4FXO , TDM400P card I am able to call outside , after modifiying extensions.conf with exten => _9X.,1,Dial(ZAP/1/${EXTEN:1}) using this , I can only dial through one of the port , Actually I want to dial outside using round - robin search After reading the manuals , I have plans to modified the above line as exten =>

Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY

In response to those who asked for a better explanation of what the Vilno software does, here's a simple example that gives some idea of what it does. LABRESULTS is a dataset with multiple rows per patient , with lab sodium measurements. It has columns: PATIENT_ID, VISIT_NUM, and SODIUM. DEMO is a dataset with one row per patient, with demographic data. It has columns: PATIENT_ID, GENDER.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, since yesterday the compiling of samba drives me mad. I'm running FreeBSD 5.2.1, installed OpenLDAP client/server and KRB5. Afterwards I got the sources for the 3.0.6-release, unpacked them and tried to configure like: ./configure --includedir=/usr/local/include/ - --with-libiconv=/usr/local/lib --with-pam --with-readline -

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 everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

Dear all, I have something about function outer() that I can't understand. Just see the following example. The two NaNs are due to 0/0, but I can't figure out the cause of the last two errors. I wonder if some one can explain this for me. ___________________________________________________________________ > sx=rbinom(10,1,0.5);ot=rbinom(10,1,0.5);ag <- rbinom(10,100,0.3);ho <-

Hello, I tried to configure a cisco 7905 IP phone using the skinny channel but I had not much luck. The relevant portion of skinny.conf is: [cisco1] device=SEP000F3487F8E3 callerid="Alex" <123-456-789> mailbox=500 callwaiting=1 transfer=1 context=default threewaycalling=1 line => 500 ; Dial(Skinny/500@cisco1) I set up the tftp server, and prepared the following

Thanks for your response! But I want to determine wheter the specific variable's value has been changed between two usages of variable, and wheter the parameter's value is changed in a function. Like the reaching definition analysis, and ud chains. 2012/8/9 ��f�� (Wei-Ren Chen) <chenwj at iis.sinica.edu.tw>: > On Thu, Aug 09, 2012 at 11:16:15AM +0800, Jianfei Hu wrote: >>

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

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

Before I start let me say I'm a relative newbie to rsync so bear with me. I've just installed version 2.5.5 and set up the following files on the server: inetd.conf rsync stream tcp nowait root /usr/local/bin/rsync rsyncd --daemon rsync.conf [www] path = /test/test2 comment = Test services rsync 873/tcp then I run 'rsync --daemon'

Hi I've been using Asterisk for a while now with the TDM400 and it seems to be working fine. I'm using version 1.2.2 and I've struck a problem when I added a Budge Tone 100 SIP phone to the network. The phone rings when calls come in and I can make calls but in all cases (internal or external calls) the other party cannot hear me even though I can hear them. I'm sure I've

Hello: I'm running R 2.7.1 in Antix (Mepis kernel but heavily debian based). I have installed libcairo2 (1.6.4-6.1). I tried installing the Cairo package since I'm interested in generating some SVG graphs, and I got the following message: Only partial output (lines with no): .... checking if R was compiled with the RConn patch... no checking cairo.h usability... yes checking cairo.h

+---------------+ +----+ ISP A | | | |-----------------| | +----+ | | +------+ | ROUTER | NET C | www | | LINUX |----------------| |

On Thu, Aug 9, 2012 at 6:57 AM, Jianfei Hu <hujianfei258 at gmail.com> wrote: > Thanks for your response! > > But I want to determine wheter the specific variable's value has been > changed between > two usages of variable, and wheter the parameter's value is changed in > a function. Like the > reaching definition analysis, and ud chains. > > > 2012/8/9

Hi, I search How To transfer call between my SIP phone. I have an PSTN line (X100P) and 10 grandstream budge tone phone. For example I want : - Reveive an external call and send it to SIP/phone1. At this point no problem. - After my receptionnist want transfert extern call at SIP/phone2... I don't known how to properly transfert call.... Thanks

I finally got my Asterisk all setup and everything seems to be working except for menu interaction between my Grandstream Budge Tone 100 and my Asterisk. I have the SIP phone setup to properly connect when pressing the 'Message' button and that's working perfectly. When the menu starts, it says press 1 to read your messages, but pressing 1 (or any number) fails to send. Does anyone

I'm not sure if this is possible, but I was hoping to find an address book that runs on Windows XP that will allow me to select a phone number and send that to my Asterisk. The Asterisk system would make the call and connect the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out there that can do that? Thanks, Dave -------------- next part -------------- An HTML

We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For