similar to: dialing out and ringing issue

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

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

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

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

Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb

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.

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

asterisk-users-request@lists.digium.com is believed to have said: >I don't mean to be rude to everyone who responded to this question, but >I think that everyone is answering the wrong question. The point is that >the message waiting indicator doesn't light up, at all, ever. All that >happens when messages are waiting is that the display blinks and the >phone gives a

I've come across an odd dialing problem with my * setup. After * has been running for a while, if I try to dial out on any of my zap channels, (both are X100P cards) it picks up the line but never sends the DTMF. Has anyone heard of or seen this problem before? Right now I'm looking at 19 hours of uptime on * itself. If I restart it, everything works fine for a little while, then the

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

<DIV><FONT face=Arial size=2>hello All,</FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9</FONT></DIV> <DIV><FONT face=Arial size=2>to take an outside call through the hardware pbx, our

I'm building a dialplan for use with a bunch of GXP2000 desk sets. During testing, we had some user issues surrounding the lack of an on-phone dialplan. Users would hit 9 and sit there waiting for a redial tone, and the GXP would time out, sending just '9' to *, which couldn't do much other than spit back a 404 or play pbx-invalid. I turned on the "early dial" option

Hello, Here's what I'd like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. I've done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), I'd like to do it with H323. Problem is, how to present a

Nevermind. Figured this out. I needed the following in extensions.conf to enable outbound dial. exten => _9.,1,Dial(Zap/2/${EXTEN:1},70,Tt) Thanks -----Original Message----- From: Info [mailto:info@psgsite.com] Sent: Tuesday, August 17, 2004 9:27 AM To: 'asterisk-users@lists.digium.com' Subject: RE: dialing out Thanks to Greg Hill for pointing me to the 'sip debug on'

Hi, I'm trying to dial out on a vonage line connected to a zap channel using stuff like: exten => 200,1,Dial(Zap/2/${EXTEN}) but it doesn't work - when i dial in the extension, i can see on a phone connected to the same line that it's gone active - but there's no dialtone. also tried adding a wait period before accessing the line and exten =>