similar to: ATT: marrandy - Re: Grandstream Budgettone 102

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

--- "Chandra" <chandra@digital.com.np> wrote: >i have been using grandstream budgettone IP phones and they work fine >except that these phones times out after some hours.. i ahve seen that >the phones working ok are next day unregistered and sip show peers do >not show their IP and although these phones can make calls , they >cannot be called. They Sip show peers

Hi! Here is the Output: bash-3.00# id NTBV+neuwald uid=5000(NTBV+neuwald) gid=5006(NTBV+dom??nen-benutzer) bash-3.00# /usr/local/samba/bin/wbinfo -r NTBV+neuwald 5001 5002 5003 5004 5005 5006 5007 5008 5009 5010 5011 5012 5013 5014 5000 5015 5016 5017 5022 5029 5033 5035 5021 5023 5019 5032 5028 5027 5034 5020 5026 5030 5024 5018 5036 5025 5031 Ciao, Bj?rn -----Urspr?ngliche

hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and

hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf entry

Just got my new Budgettone phone, and I've got a couple of issues. Most important, it doesn't seem to be querying for the time via NTP. I put a sniffer on the line, and once it boots up the only outbound traffic it generates is an attempt to contact a TFTP server, which is programmed in as 192.168.0.168. . . Must it first find a config file (it's asking for "cfg.txt")

Since the last firmware upgrade we've been experiencing some odd CallerID behaviour. Instead of the LCD showing the calling party's #, the phones are showing the internal extension being dialed. This is probably a really stupid fix I'm overlooking, but I was hoping someone could offer some insight. Thanks! -Corey -- Corey S. McFadden McFadden Associates - Technology

Hi all, I'm quite new on this mailing list, and I discover the asterisk world. I m experimenting a PBX with SIP phones, grandstream budgetone (not expensive for tests) All the features I need work just not one : the supervised call transfers. I know there are a lot of posts about that, but none gave me the correct answer (unless I missed it). Here is my config : 2 sip phones BT102 with

I have my new phone mostly working. I do have a couple of residuals that I cannot find mentioned in the list archives: 1. Is it possible to set the volume in these things? I hope I didn't miss it, but I've looked in the doc, the FAQ, and the asterisk archives and don't find anything. The displays in the pictures all have more bars on them than my phone does, and I need a bit

Hi, Quick question to all the electronics gurus out there.. I unpacked my second GS phone yesterday (had it for about a month!) and set it up.. This morning the power supply is dead.. I have looked for a new one online (In the UK using Maplin let me know if you know a better place.) becasue it would probbaly take too long to get one sent from China or the US and I need to get that station

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

Looks like Mozilla forgives that problem you are talking about too.. I have had no problems configuring the GS phones using Mozilla on RH9.. > -=> On Sun, 24 Aug 2003 09:47:06 -0400 (EDT), John Vozza <john@netrom.com> said: > > > Just noticed that version 1.0.3.81 has been released on the > > Grandstream website. > > Doh. I see they still haven't decided

Hi, I am thinking about getting some of these phones to use with my Asterisk system.. So I would be interested to hear from the people who have tired and tested these phones.. What is the voice quality like using alaw/ulaw or G.729?? Are there any problems getting them to play nicely with Asterisk?? How sturdy is the construction (reciever, handset, buttons, cable connections, etc..) ?? How

Hi all, can any help with our strong problem: we are using Samba on Linux as fileserver for Win2k clients running CAD software "Solid Edge". This program opens many files and on big projects freeze. We tried configuration: - samba 2.2.3a or 2.2.4 - RH 7.2 or 7.3; kernel 2.4.18, 2.4.19 up to -pre10-ac2 - i686/1GHz CPU, 512 MB RAM - "/proc/sys/fs/file-max" from default 8192 up

Hi, I am interested in the thoughts of the people who know the architecture of Asterisk quite well.. What I am interested in is the performace difference of Celeron vs P4 vs Xeon.. I know Intel marketing says that servers should be Xeon, Workstations should be P4 and desktops should be Celeron and they are priced to the specific target market.. I am on a tight budget and so I am looking for the

> > I guess my big question is: is it possible to have extensions mapped to > > people, not to phones? > > Yes, you just need to link the user/extension to a technology/channel > when logged in, and to a bogus value when not so that your dial will > fail quickly and fall through to voicemail. Also you will want to make > sure your voicemail goes out in email since you

Hi, I am running Redhat, I loaded the mysql and mysql-devel RPM's and then recompiled *.. I thought it would be that simple but it looks like I have missed something becasue it doesn't look like the module has been complied.. What did I leave out? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

Hi, First of all, thx for fast answering. > I hope that the above commands are really right, because you said folder_a but the name of the folder is "folderA". Sorry, the commands are right :) > The all other things include the groupmaps? What do you mean... when i uses the command "./net groupmap list", i get the following back

I sent this to freebsd-questions earlier via the newsgroup, but I just found out today that the list-to-news gateway has been down for half a month now. Any idea when it's going back up? Who was hosting that service anyway? And now for the question.. I have a test server that is being used to test a 4.4-REL to 4.8-STABLE upgrade, and I've found the system will no longer boot unless I