search for: bugeton

Hi, Just got a couple of Bugetone's yesterday.. I updated the firmware to 1.0.3.72 but now it seems to have problems getting the time from the default NTP server.. I also tried time-b.nist.gov but still the time does not seem to work.. I am able to contact the time server from my PC.. Anyone got any ideas? -- _____________...

...Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian Capouch Sent: Thursday, January 13, 2005 5:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream Bugetone 101 & mwi dean collins wrote: > > In addition I currently have a $2,000 bounty for a video version of > Meetme and haven't had a single person even reply to take up the > challenge. > > How much more encouragement do you need? > One of the hardest things for trad...

how do these phones perform? ok for office use? work well with asterisk? any info would be appreciated.

I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald

asterisk-users-request@lists.digium.com is believed to have said: >> Haha....well the MWI is the blinking blue LCD. The message button >> is "reserved for future use" Hang in there. There will soon to be some >No the message button call the number you configure in the web >interface. Presumably voicemail, but could be your mistress. Then it's easy to get

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

The process for transfering a call with the Bugetone is as follows.. 1. Press "transfer", you will get a dial tone.. 2. Dial in the extension to wish to transfer to.. 3. Press the "Redial".. (on the newer phones this is the "send" button) You don't need the "t" option on your dial string to do transfers...

Has anybody got the Early Dial feature working on Asterisk with a grandstream phone ? I can do two digit dials eg 12 and it works fine. When I press a 3rd digit I get a busy response. I did add the auth=plain text in my sip.conf file but to no avail. I have also done the redirect line to make UDP port 0 redirect to port 5060. Anyone had any success ? Cheers Nigel -------------- next part

Hi everyone, I know someone makes a product that's a POTS phone to SIP converter, where you just plug your POTS phone in one side and the network cable in the other. Has anyone successfully used any of these with Asterisk, and if so how expensive were they? I ask partly out of frustration with the FXS cards but mostly because it would make installation MUCH easier for what we're

...tant to push * as a solution for this client, mostly due to my limited knowledge of * and its features, I am concerned that 1.) The phones I chose will not fit their needs, and 2.) * will not provide the features that they need. Things like shared line appearances. 3.) Phone Compatibility; The Bugetone 2000 has been highly recommended, but I'm still looking at Polycom (501) and Aastra phones (9133i). How are your experiences with these phones? Is there a better model for supporting advanced features? Features like FollowMe, Share Line Appearances, etc? Being my first install, should ther...

Hi again. I cant get my Budgetone registered in Asterisk, and I cant find what's wrong... uff. This is my config: This fragment is from my sip.conf: [12345] type=user user=12345 username=12345 secret=12345 authuser=12345 qualify=1000 nat=no host=dynamic dtmfmode=rfc2833 reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw context=sip_default And this is from my