similar to: Quick summary of Grandstream survey results

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 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- I am attempting to setup my Budgettone phone for use with my * server and am having problems obtaining an IP address. I have checked the phones settings to make sure it has dhcp enabled and it is. The display says no IP. I bought the phone but do not have any documentation other than the Wiki, but I am still at a loss. What could be preventing the phone from picking up an IP address? Any

hi .. i've just converted myself back to a newbie by trying to experiment with some new stuff .. I have connected two grandstream Budgettone 100 phones to my asterisk, and trying to experiment with them .. I am trying to get into the asterisk sample basically .. when I dial 1000 asterisk receives the call, but I do not hear any sound on the phone. Dialling from phone to phone direct (via

Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I

Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things

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, I have used the Budgetone 102's extensively on Asterisk and found then quite reliable as long as you update the firmware. The GXP2000 is quite a mess at the moment as the current firmware does not support 3 quarters of the advertised functions and codec support is extremely limited. I have tested the unreleased latest firmware update for the GXP2000 and it's an incredible difference

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

--- "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

I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan

[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

On Tue, 2003-12-23 at 19:22, Brian West wrote: > I'm just double checking.. I was told it wasn't possible but i'm going to > ask just in case. > > Can you set outbound callerid on a channelized T1? > >I think there is a way to do something like DID with the 4 digits of >DTMF passed before the call. It is unlikely though that you will find >someone interested

The about-to-be-released Grandstream firmware now supports multiple ringtones, but (so far) I haven't been able to unearth any documentation as to how one uses them. Anyone out there know anything about this? I've googled, read the firmware READMEs and combed the GS site without any luck. Thx. B.

How many of you are using Rails: 1. As the primary development tool at your Day Job 2. As a small part of your Day Job 3. Not using Rails in your Day Job, but are using Ruby 4. For OS/outside work/hobby -- sam http://www.magpiebrain.com/

I have just successfully got the TFTP file remapping to work such that I can have unique ringtone files for each and every extension. I added the following to my server_args line in the xinetd configuration for TFTP: -m /home/asterisk/grandstream/ringmap.cfg Now the entire line reads: server_args = -v -s /home/asterisk/grandstream -u asterisk -m /home/asterisk/grandstream/ringmap.cfg (There

If you wan't to create a ringtone with makering.pl for firmware 1.0.50, be sure to create it as ring.bin and then rename it to ring1.bin / ring2.bin or ring3.bin. This seems to be the only change between the format from 1.0.4.68. Regards, Maron

Hello all, Over the weekend, I setup and linked an Asterisk box at another site to the Asterisk box here. The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100 phones. The phones at the other end are Grandstream BT-100 SIP phones. The Cisco phones run SIP 7.1 (upgraded last Friday from 6.1), the Grandstream phones run 1.0.4.68. Both Asterisk boxes are running stable CVS

well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my

On Thursday, August 24, 2017, 1:50:13 AM EDT, David Winsemius <dwinsemius at comcast.net> wrote: > On Aug 22, 2017, at 11:31 PM, jagan krishnan via R-help <r-help at r-project.org> wrote: > > Hi all, > This is Jagan.i have been provided a task of analyzing sales data of a company in R programming...Just wanted to know,how can I pull Tally 9.1 software data into R