similar to: Budgetone 101 phones ? SIP through NAT ?

Hi, Some time ago, there was a discussion about the inability of nortel meridian pbx to dial analog tones thru an meridian ATA, and the work arround was to enable 808 function that makes the dtmf tones long for the current call. The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port. Anyone can say me who to actually use that function (you dial something or is pbx

Well, I don't know how to tune it more, it connects at about that rate in a mediocre rural landline. ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be using more of the necesary scarce bandwidth AND dropping sample info in each frame, thus making audio choppy and unclear. Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for it in

Hi! We planing to by some PRI/BRI equipment to replace our existing telephone system. So I am going to try this: ISDN Card Outgoing Digital Call / Capability: Unrestricted digital information -> octobri -> asterisk -> TE410E -> Internet Provider / Receiver for Capability Unrestricted digital information The question is: Is asterisk possible of transmit this digital call to the

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can

After searching the archives, I came acrross a few people mentioning this, but I never saw anything about what became of it. Has anyone tried to make a virtual modem that could be directly handled by astrisk, I saw a while ago that someone was going to try and make one using the same DSP libaries that the WinModems use, but then nothing. Would do this even be possible, and if so, what kind of

I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32 C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.20 25 Jan 18:25 Error Aborted 192.168.16.32 C:\Program Files\TFTP

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully

Hi Folks, Since there isn't a grandstream forum AFAIK I guess someone here may be able to shed some light on this. Apologies if this is viewed as offtopic.. I think I may have killed the firmware on my Grandstream Budgetone 101. I found a source for the 1.0.5.30 firmware and made the files available over tftp. The phone downloaded the files but now doesn't boot and hangs with a blank

I just got a Budgetone 101 and I have it hooked to my * box. I thought I'd read somewhere that we can program the buttons on these phones to send DTMF tones, thereby effectively programming them. However, according to the user's manual, they have predefined SIP functionality. My dialplan implements the festures I want (transfer, message, stuff like that), so for uniformity, I'd just

Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? Best regards, Daniel ANDRE -- Daniel ANDRE (mailto:daniel.andre@iris-tech.fr) IRIS

I know there are several folks hanging around the list that resell VoIP products, I'd be interested in getting your info off-list as I'm trying to persuade someone to get me a budgetone 101 for Christmas ;) I've checked the yahoo site, and they seem to only carry it in white.. Ideally, I'd like blue, so if you have it, please let me know. Also, I've been watching the list for

Hi, not sure if it is against the rules to sell second hand equipment in here but haven't seen anything against it so here it is. I'm upgrading to 2 lines so I have some spare equipment for sale here. This is an ideal starter pack and will get you going with 1 line and 1 extension. 1 x http://www.digium.com/downloads/product_sheets/X100P.pdf 1 x

Maybe I'm loosing my mind but I've just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. I'm using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone I'm missing? Cheers, Dean -------------- next part -------------- An

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

Hi there, I'm having problems with the Grandstream Budgetone 101 on hangup - "show channels"/"show channels concise" output is still showing the call's channels as active. The problem does not exist when I use SJPhone, so I'm assuming it isn't an Asterisk configuration issue. Has anyone seen this, or better, does anyone have a fix? :) Thanks, David. --

Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busy I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones

Hi all, I'm having a really frustrating time with a bunch of BT-101 phones. They've been trouble-free and working very well for the past several months. A couple of days ago, some of the phones (but not all of them, yet) have started acting very strangely. All phones are running firmware 1.0.6.7, and are identically configured (except for the user/authenticate/password things) on both the

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Yes and No The D500 is a terrible thing First problem: it sends some horrible DTMF, so if your voicemail is configured to send #H and *H, it will not work, configure it to send numbers, like 8H and 9H (H is a placeholder for the extension). I also managed to use the MWI (message light), it's a perl script that is in voip-info.org, but with a little correction because the wiki distorted it. If