similar to: Budgettone Newbie

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, 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

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'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

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.

Akemi Yagi <amyagi at gmail.com> wrote: > This issue? > > https://bugs.centos.org/view.php?id=17767 This is a huge bug! Using Firefox to watch videos is a basic activity. The bug with grub2 a few months ago was even more serious: it made systems unbootable. It's discomfiting to see two major bugs so close to each other. -- Yves Bellefeuille <yan at storm.ca>

--- "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

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

I upgraded from RC2 last night, but have a major call quality issue. Heres our setup: 1 FXS and 1 FXO card. Incoming/Outgoing calls via IAX trunking from our provider. G729 running between us and the VoIP provider. Two handsets, one BudgetTone 102 and a Cisco 7940G running the 7.2 SIP firmware. Both these phones are using ULAW to the server, and we have plenty of G729 licenses on the server.

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

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

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

Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN

I'm sure there must be a setting I'm missing somewhere, so I thought I might was well ask here. Conversations are punctuated by sudden replacement of a given syllable or so of conversation with a DTMF tone. I would hope perhaps there's some kind of setting that has to do with the way it detects inband DTMF? I'm pretty sure it's an artifact of this particular ATA; my

I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. This problem began for me over a year ago, and continues up to the latest versions I've installed (1.6.2.13). It happens randomly, and the suggestion on one of the bug tracker tickets that it is instigated by a small network leg looks to be on point to me,

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

Like most others on this list I also have some really annoying echo whenever a call goes out to the PSTN from a SIP phone... SNOM/Budgettone -> Asterisk -> X100P -> PSTN I have tried every echo canceler in the makefile and turned on and off aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and I can get it reduced to only a few seconds on the intro of the call and

[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