similar to: Transfer from Grandstream BT100?

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

Does anyone know of any reliable supplier for Grandstream phones? I tried dealing with David Li from Grandstream, but after emailing him an order in August, and asking how he wanted payment, I never got a reply... James Ho from DGTimes was happy to give me pricing, but when I sent him an email asking for shipping costs, I never got a reply... I tried dealing with John from Chagres Ventures, but

I'm trying to get call parking to work, but I've run out of things to try. I can place a call between two internal extensions, then on one extension transfer the call to extension 700, and the call gets parked on 701 but I don't hear the extension number when I do the transfer. I can hangup and call 701 and get the call back. Here's what I see: (comments added on lines starting

Hi Guys, Sorry about writing to that list but could not find better place. I have Grandstream BT-100 phone, btw, was working great with Asterisk. I have upgraded the phone, and during upgrade something went wrong. Right now when I power the phone I can only see some garbage on the LCD display. does not react on any buttons, pings,..... Maybe somebody has any idea if it is fixable or I can just

hi all i upgrade a bt100 phone and it can't resgister with asterisk Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request: Registration from '<sip:@172.16.4.249>' failed for '172.16.4.226' is was working with the version 1.0.5.3 some bady now what is hapening? thanks in advance Rodney

Hi, I've had a bunch of people asking about purchasing GS phones. I'm happy to announce that we now have an online purchase system available. We have: BT-101 BT-102 HT-286 Plus the full line of Digium hardware. We will be adding other products in the near future so check back :) Buyers are responsible for s/h costs. The URL for the phones is:

Hi All, I'm sure this is something simple that I have missed somewhere. When I make a call using BT100 over IAX2 with Voipjet terminating I don't get a ringing sound whilst I'm waiting to be connected. The destination party can answer the call (they do get ringing) and conversation can take place. I don't get this problem with X-Lite softphone? Any help appreciated -

Asterisk Users In an attempt to help Asterisk move forward, a number of us have decided to create a book. It would initially be released as an "ebook" that could be sent to newbies to help them up the rather steep learning curve. Ultimately I would like to see it published and sold in bookstores (preferably by O'Reilly & Co.). Below is the outline for the book. We REALLY

Has anyone tried installing * on a box with two eth interfaces which is acting as a NAT box? I have only one IP at this point and I would like to get * working without all of the NAT issues. My idea is to run * on my gateway (which is also running the firewall and masquerade services). All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the NAT screen, and will connect to the *

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

I have two problems with Asterisk and my Budgetone 100s. 1. Sometimes when outside callers dial my Budgetone it will ring twice, then the caller will receive a busy signal. This happens intermittently and I cannot pin down the circumstances of when it works and when it doesn't. Asterisk logging doesn't give me too much info when this happens: *CLI> -- Executing

Sorry to everyone on the list, but for some reason this is the only reliable way to get hold of John. John Brown of Chagres Technologies, please contact me! I have been trying for weeks now to get hold of you via email and phone after wire transfering money into your account for the Grandstream phones we ordered, but so far I have not had a single response, nor have the phones arrived! Please

Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware. http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch= -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Hello, Has anyone gotten a Grandstream BT100 to work with Level 3's 3Tone? I've been able to get my extension to interface with it, but there is no sound and the call on the GS side terminates prematurely. Here is the relavent portion of the SIP.CONF [4007] ; Budgetone BT100 type=friend insecure=yes context=test-budget username=4007 fromuser=4007 callerid=4007 host=dynamic nat=yes

Hi everyone, I'm trying to get early dial to work with our grandstream 100 phones. The phones use SIP, asterisk is 1-0-5 on debian GNU/Linux (sarge). Outside connections are via 2 ISDN BRI (British Telecom) lines using 2 billion isdn cards and bristuff. The phones are set up to be in context [internalphone]. I numbered all the internal extension with _6XX That works well with early dial.

As I've said several times on this list[insert usual apology here], I still haven't received the last 20 of 100 phones I ordered over 2 months ago. If you get a hold of them please let me know MATT--- -----Original Message----- From: mikeu [mailto:mikeu@execpc.com] Sent: Saturday, January 10, 2004 12:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Chagres

Hi, after upgrading to Asterisk 1.2.8 from 1.2.7.1 I got a problem with Grandstream BT100 after making an attended transfer (FLASH + NUMBER + SEND + WAIT ANSWER + TRANSFER). After the transfer, the display clears all the info except the clock, there is no dial tone, the WEB admin stops working. Incoming calls make the display light turn on but there is no ring and no callerid on the

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

If anybody is interested, I have an early version of my Call Manager for Windows application integrated with Asterisk. CMW is an application bar (like the task-bar) that docks to the top of your desktop window. It provides the following functions: 1. View Call-Related Information (Caller ID, Call State, Call Direction) 2. Monitor Status of Asterisk Stations (Channels) -- BLF or "Busy

I have the following: Call -> PSTN -> * -> GrandStream 101 1.0.3.81 The GS displays "ohn ro n2600" when the call is past to the GS. If I pass the call to a XTEN client, Caller ID shows up. Any ideas ??