similar to: Gradstream Budge Tone-201

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; 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

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

Hello, I have a problem with one of three [topic] phones. The phone, which is on the LAN in the same subnet as Asterisk, keeps unregistering from the Asterisk server. Whan it is unregistered there is no way to make a phone call from it, but once it is rang by any other of the phones it registers to Asterisk again. The other two are absolutely fine. The problematic one [ecco] puts this messages

Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat

hi, Several doubts aroused after I read through all the information provided in former mails. They are > > >> > > I think this would be a great project. However, I would rephrase it > > to be more concrete. > > > > How about taking a linux distro like redhat or gentoo or whatever you > > are familiar of comfortable with, and try compiling the whole

On Sun, Mar 30, 2008 at 11:34 PM, Kumaripaba Miyurusara Atukorala <paba50 at gmail.com> wrote: > 1) I thought of taking the gcc compiler and compiling it with llvm since it > is easier to make test cases to test the system. Is gcc compiler already > built with llvm? if so I have the linux kernel as the second option. What is > your openion on this ? We rutinely compile linux

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the

Thank you and sorry...There is something going wrong with the system I only sent one mail... _____ From: Kerry Garrison [mailto:kerryg@techdatapros.com] Sent: Wednesday, May 11, 2005 5:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [SPAM] - RE: [Asterisk-Users] Grandstream-Budge tone - Email found in subject This is usualy a problem with either

Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed

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

I am using Icecast2 with Ezstream, sending a playlist of mp3s. The signal quality is fine over my home network; but out on the internet it is poor, with frequent interruptions and losses of connection. I would like to know which of the following two conditions, if any, can be causing the problem. First, I have a dsl internet service, with only 160k upload capacity. Second, my home network is

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

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

Chris Lattner wrote: > > On Mar 29, 2008, at 11:53 PM, Kumaripaba Miyurusara Atukorala wrote: > >> hi, >> This e-mail is written to involve some of the project ideas in LLVM >> in GSOC this year. >> I was looking in to the ideas mentioned under improving current >> system and found the idea of "Compile programs with the LLVM >> Compiler" to be

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

I have a quick question. I am using Asterisk 1.4. I have a user that has changed phones (grandstream budge tone 200 to a polycom 330). I have changed the sip.conf and extensions.conf. I have also unplugged the old phone and plugged in the new phone. I get the ext showing on the phone, but when I do a sip show peer 5000 the old ip address and phone show up. I did a sip reload and a dialplan reload.