similar to: Grandstream Voicemail

Hi everyone, I just got a Sipura SPA-1001 and have connected my Tivo Series 1 (yes its old). When I do a test call with Tivo, the call always fails (it seems to dial the number but never connects). I can pick up the phone line and hear the Tivo "talking". I've tried looking around for anything special I need to do but its still not working. I can connect a phone to the SPA-1001

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 see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip

G'Day List, Can someone help me out a bit please. I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying to install * I am following http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation After running: cd /usr/src/asterisk make clean make make install make samples The instructions says: Configuring Asterisk - Login to your server as user

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

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

I'm sure this has been asked more times than anyone cares to count, but I want to make sure I get the right stuff. I'm installing Asterisk at home solely to play with and learn VoIP (plus it sounds pretty cool so it has that going for it too). Eventually all my phones (~4 of them) will go through my Asterisk system. I've ordered my server hardware, here's what I've got:

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

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

G'Day All, How do I fix this: I receive a call at the extension. Press the hold button. Music on hold starts. When I place the handset back on the cradle, the call gets hung up/disconnected. The Phone is A GrandStream Budge Tone 100. Thanks

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

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

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.

Hello: Hoping someone might know how to resolve this (probably an easy one). I have one Asterisk PBX with a single NIC and an FXO card with PSTN line attached, and one IP phone (Budge Tone 100) on the LAN. Via the phone I get no dial tone, and dialing 9, <number> doesn't allow me to dial out. I also need the phone to ring when the asterisk PBX is called. I have modestly tweaked the

Le 26/04/2020 ? 08:31, Gordon Messmer a ?crit?: > Are you sure the input lacked a space character, and that this isn't just a > font rendering bug? 100 % positive. I tried it on a different sandbox server without X installed, but only with xorg-x11-xauth. >From my workstation running OpenSUSE Leap 15.1, I connected to my admin console using the following command: $ ssh -X

Hi, I'm using the current version of Ubuntu, and the Wine version the system down loaded. I'd like to use my photo application "Fixfoto" but for some reason I can't get it to work. I can use some apps e. g. XnView, or IrfanView without any problems but for some reason I can't get Fixfoto to work with Wine. I've installed Fixfoto using Wine but when I click on the icon