similar to: Supervised transfer problem with BudgetTone

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")

--- "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

[Posted here becasue your mail server is rejecting my direct reply to you.] Hi Martin, AFAIK SIP can run on both UDP and TCP but I have only seen it used over UDP.. :) To setup the GS phones you need to open up the following ports (If its still set at the defaults)... UDP/5060 UDP/5004 UDP/5005 UDP/5006 UDP/5007 I have not tested the GS phone through a firewall yet but this config should

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

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

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.

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''m learning Ruby on Rails with the Agile Web Development With Rails book. I have installed InstantRails 1.4 (under windows xp 2) and after doing all steps (gem updates, rails demo, ruby script/server which launches WEBrick server), I can''t access the localhost:3000 with the browser. I tried 127.0.0.1:3000 in apache config, then with my IP address 192.168.1.100:3000

Hi, i try to etablish unencrypted onnection and have in my config: Cipher=aes But this causes the following error: 2018-05-23 12:08:27 tinc.backbone[14746]: tincd 1.0.31 starting, debug level 5 2018-05-23 12:08:27 tinc.backbone[14746]: Got fatal signal 11 (Segmentation fault) 2018-05-23 12:08:27 tinc.backbone[14746]: Trying to re-execute in 5 seconds... I have no idea what i'm doing

Strange problem and I'm not sure what's causing it. I'm using IMAP. A new message arives in the inbox. I see it displayed in the message list in bold. I click on the message and it looks like it's reading it by the previous message that was in the window remains and the new message is still bold (unread). However I can then click on an old message and then click on the new

Hello, I'm having some * and sendmail integration problems, probably because i don't know too much about sendmail. My server crashes when I forward voicemail from one * voicemail box to another, everything else works. E-mail notification works on all boxes when new mail arives, the problem only seems to occur during this forwarding function. It's a difficult problem to

Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am

Hi- I'm at my wit's end trying to get a Grandstream BT-101 POS to register on my * server. Asterisk version is 1.2.1. GS Firmware is rev 1.0.6.7. Basically, I've setup the phone following the instructions at voip-info.org, and it registers for about 10 seconds, then after receiving the SIP NOTIFY from the * server, goes into "flashing display" mode, which indicates some

Luca: 1. We are not a consulting service. We *help* with R pogramming issues. Users are typically expected to make an effort by providing R code and, if appropriate, small data sets that illustrate their difficulties. 2. SEARCH! e.g. on "text processing R" or some such; or try Rseek.org with such searches. R has extensive text processing capabilities, e.g. via regex's. 3.

Hi. I was wondering if there's a way to do supervised transfers on the budgetone 102 . Blind works ok, but can't do supervised. I thought that with the flash button that could be possible, but seems I'm wrong .... matteo.

I've seen a lot of traffic on the list recently about which phones can do supervised transfers and which cannot, and I have to admit that I'm a bit puzzled. Our existing PBX, which is software based, handles the transfer functions for our call center -- the agents never touch their phone, and instead use software. We can plug any old phone into it, and it'll work just the same. So

Is there a way to keep the incoming CallerID from the PSTN and pass it onto the sip phone receiving the supervised call transfer? The receptionist receives the PSTN callerID, performs a supervised transfer, we get her local SIP callerID, not the original callers. The main reason we would like the true callerID is for asterisk monitor to name the file correctly for call records. Is this

Hi all, I'm trying attended transfer on Asterisk 1.0.7 and AT-320 phone. I met a lot of problems during this steps, while in the blind transfer all works fine. I had this kind of problem: CASE 1: A call B B set on hold A B call C (that is busy for some reason) B try to get the first call with "hook flash" (or pressing the "hold" key) and A stop to work. B