similar to: video phones on 1.4.7

hi, do you someone test this http://www.grandstream.com/y-gxv3000.htm? video works? (it's have H264 video codec) i want this topology gxv-3000 - nat -{Internet}- Asterisk -{Internet}- nat - gxv-3000 --------------------------------------- Marek Cervenka LCNA - http://lcna.slu.cz =======================================

Well, the recent talk about video phones and a project I've had lurking which has come to the top of the pile recently made me go out and buy a pair of Grandstream video phones. And stack-me-sideways they "just work". Amazing little boxes too (actually not that little with a 5.6" screen!) They even have a web browser built-in with lots of "stuff" in the config

Anyone using an N900 with asterisk yet? Had mine for a while now and VoIP (voice) has been working really well, but the new firmware update brings SIP Video calling too - so just given it a go... The fly in the ointment is that I only have a Grandstream GXV3000 to test it with ... And it worked - sort of. If I call the N900 from the Grandstream it's OK (H263 and H264 seems supported),

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial("Zap/2-1",

Hi, I needed to create config files for downloading to Grandstream devices and made a little script for it. It seems to work fine for the HT486. The script needs a config file (cfg.in) which is in this format: P2 = blah P10 = hrm ...etc... The configfile may contain comments (starting with '#') and empty lines. Mind that the 'gnkey=0b82' shouldn't be in the configfile, as

Hi, everyone ! Looking at this explanation : "When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the

Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a grandstream registering to asterisk, named sip0. Sip0 registers, via sip, to another asterisk box, sip1. When I place a call from the grandstream, it will travel through sip0 to sip1, where it is then placed to the PSTN. Nothing can reinvite, this path is

Hi, One of my clients, wants to use * box to run weekly meetings between remote locations over the internet. What would be the best configuration for this? We are talking about two conference rooms. I am referring to the actual hardware/software and bandwidth requirements for this to work well. I have run two software video phones and I had marginal results with it when displayed on large LCDs,

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hi, I install * and is working fine. I have 3 FXO cards w/ 3 phone lines. All the phones are SIP phones (Grandstream). The SIP phones from the same LAN w/ Asterisk are working but on the external phones (from the Internet) I don?t have sound. All the Grandstream phones from the Internet are register from different locations behind a NAT. All the sip users are register on * but the main issue is

Hi, recently I changend a few things in the configuration of the Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that different groups of SIP-Phones are using different trunks to the outside worls, so I moved some of them to a Support context. However, dial out from this phones failes as they're still looking for an extension in the default context, which doesn't

Hello guys, I ask you to share your experience with your BudgeTone 100.... I have my asterisk @ work and I've bought a GrandStream BudgeTone (SIP phone) and I usually use X-Lite I have plugged my BudgeTone into my home network because I want to be called even at home. I succeed to register my X-Lite with Asterisk from home but I can't do that with my BudgeTone. (I don't know

Hi, we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One GrandStream GXV3000 is used for the tests. He is registered to asterisk 1.6.2.18 asa well as 1.4.35. Calling echo test is OK on both servers, get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP trunk from both others servers is also working well. What fail, is video on echo test from asterisk 1.4.42

Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

Suppose I have three numbers p1, p2, p3 with 0 <= p1,p2,p3 <= 1 and p1+p2+p3=1, and a function f=f(p1,p2,p3) = f(p1,p2,1-p1-p2). How to draw a contour plot of f() on the p1+p2+p3=1 plane, that is, an equilateral triangle? Functions triplot(), triangle.plot(), and ternaryplot() give only scatterplots, AFAICS -- Robin Hankin Uncertainty Analyst National Oceanography Centre,

I have 3 vectors: p1, p2, and p3. I would like each vector to be any possible value between 0 and 1 and p1 + p2 + p3 = 1. I want to graph these and I've thought about using scatterplot3d(). Here's what I have so far. library(scatterplot3d) p1 <- c(1,0,0,.5,.5,0,.5,.25,.25,.34,.33,.33,.8,.1,.1,.9,.05,.05) p2 <- c(0,1,0,.5,0,.5,.25,.5,.25,.33,.34,.33,.1,.8,.1,.05,.9,.05) p3 <-

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I have used the handytone 488 from grandstream in the past.... However I need to be able to send a number to a unit like the 488 and have it dial out. Is there a unit like this available? Basically a 488 unit that can place a call out. Jerry

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.