similar to: Budgetone + NAT: Firmware Version?

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 have spent the morning on this project, still without success. Summary: Yesterday I inadvertently unplugged my Grandstream phone. I might add I did a rebuild of my s/w from CVS at the same time. Since then, the Budgetone seems to talk SIP just fine, but the RTP being sent to it by asterisk "doesn't make any sound." It was suggested I do a factory reset of the phone, which I

Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ

I have seen two references today (don't recall whether here or on one of the other VoIP lists I read) to people having the .78 version of the firmware installed on their phones. I'm keen on getting hold of it, but their support page still offers only the older version. Is it out there somewhere, and how might one have at it? Thx. B.

Hello, I have a problem with asterisk and Grandstream BudgeTone-100. With default configuration everything works (in anonymous mode and fixed IP), but if Im trying to enable registering, it dos not work. I used 'sip debug' and verbose level 10, nothing happens if I switch telephone on (no messages about bad auth etc). As I understood, after switching phone on at first it will try to

Hi All I have 3 GrandStream BudgeTone-100's which connect to an * box with a HFC-S based ISDN card using ISDN4Linux. I have setup the BudgeTone-100's to use Early Dial which for calling between the three phones works well, but for the external calls using the following extension exten => _9.,1,Dial(Modem/g1:${EXTEN:1}) Only the first digit is dial on the ISDN Line. Does anyone know of

Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my "POTS backhaul." On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig

Just noticed that version 1.0.3.81 has been released on the Grandstream website. Have fun... John ------------------------------------------------------------------------- NetRom Internet Services 973-208-1339 voice john@netrom.com 973-208-0942 fax http://www.netrom.com -------------------------------------------------------------------------

I have a couple of Budgetones that I am playing with trying to get them to work with * from a remote network over the Internet (yes NAT joy!). My * server is in my DMZ and I have 5060 and my RTP range forwarded (UDP) to my public address (through a Cisco PIX). Internally, I can setup my budgetone, it registers and works great. I then have a Linksys router connected to another Internet

I'm getting 1.0.4.30 I think it is, in new phones, but all that's on the website is 1.0.3.81 Where do you download newer versions ? And, will anyone else's firmware work on these ? This firmware seems to be flaky at best. These Budgetone phones SUCK with NAT involved.

Hi. Has anyone experienced with the new firmware .77 ? There's Day Light Saving time now, but haven't time to play with it, till now. Matteo. -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : mbrancaleoni@espia.it

I wonder if someone out there could loan me a peek at their sip.conf? I have conflicting advice, for instance, about whether or not to use "nat=1" and also whether or not the ATA should be registering with the instance of asterisk it is going to be using to dial out. Thanks in advance. B.

I'm using Asterisk on a open server (no firewall or NAT) and trying to communicate with a Grandstream BudgeTone 102 SIP phone which is behind NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS about a week ago. My problem is that I'm only getting half-duplex communication -- I can hear voice from the Asterisk server but the server does not understand any voice from

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,

I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process. At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call

Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are "generic," and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the Budgetone and any other phone, the call is setup and sounds good. If I hang-up on the Budgetone, everything is ok. If I hang up on the other phone, the Budgetone