similar to: Sipura 841 vs Grandstream GXP2000

I've searched the Wiki and this forum with little success. I have a TDM400P in my server which functions fine. Except it will continue ringing about 3 times after hangup. I.e. it's failing to detect the hangup tone. I was previously running a Sipura 3000 and had the same issue. After researching and some timely assistance I was able to determine the hangup tones applicable to

Hi, Not sure if this is OT or not - apologies if it is. I'm looking for an asterisk consultant in the Sydney area to help with the rollout of a couple of Asterisk setups for clients of ours. If you're interested please reply to me /*off-list*/ at tonyd -at- zeroeffortnetworking -dot- com -dot- au (fixing the address of course :) thanks, tony Zero Effort Networking Pty Ltd ABN 38

Apologies if this has already been posted to the list. I sent it about an hour ago and it hasn't shown up yet. I've changed the format to plain text in case that was the issue. I'm having a major issue whereby voice is going out fine but none is coming in. If I ring someone I will hear them answer and then nothing. They still can hear me. Looked at the logs and with my

Hello all, I was wondering if anyone know where i can find a book on Asterisk, i have been told about VoIP With Asterisk but i am unsure where to find it, any ideas plase?

Hi i use a lot of Grandstream GXP2000 with BLF How to set up on the same key BLF blinking call interception? So that someone is able to take a call that is destinated to another user phone Thanks bye

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to

We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

Version - Asterisk SVN-trunk-r12793M (1.2.4) I have 4 Grandstream GXP 2000 phones configured. However at the moment, I have had to disable BLF, Hints, and Call Limiting due to an extremely annoying bug which seems to make the phones channels "lock" in busy after a call has been hungup. If I do a show hints after say extension 200 has hung up I get the following -= Registered

Asterisk 1.0.7 Has anyone got the hint function working, and maybe with the GXP2000. I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment trying to get the LED's to light up. On ext 690, button 1 is setup for ext 691, I did this using both methods 691, and <sip:691@192.168.69.1> On ext 691, button 1 is setup for ext 690, I did this using both methods 690, and

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

Does anyone have a solution for the problem of file and directory names containing characters which are OK in DOS/Windows but illegal (or at least unwise) in Unix? I have Windows clients who have written directory names containing "$" characters to Samba shares on a Solaris box. This is fine for the clients, but plays havoc with Solaris programs needing to access the files, eg. backups

Just got my (10) BT102 phones, flashed them to 1.0.5.20 and all work. No duds at all. Not a bad little phone at all. Doug

Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? TIA Bruce Komito High Sierra Networks, Inc.

I just grabbed 1.9.18p2 and tried compiling with gcc 2.7.2.1 for a Solaris 2.5 box, but it didn't get too far - after a bunch of macro re-defines (eg. "/usr/include/sys/termios.h", line 32: macro CTRL redefines previous macro at "/usr/ucbinclude/sys/ttychars.h", line 65 "/usr/include/sys/termios.h", line 150: macro CINTR redefines previous macro at

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

Hi all. Newbie to the list, been using VOIP with Sipura & Grandstream hardphones for a few years, via a VOIP service provider (who I won't name here). I haven't stepped up to running my own Asterisk box yet, because of poor reliability of our Internet connection during non-business hours, but I'm considering it in the future. I can't provide a ton of detail about the

FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC -> HP Switch -> Asterisk : actual throughput measured at 94.1 mbps. PC -> BT102 -> HP Switch -> Asterisk : actual measured at 8.86 mbps. The

Just added a new Grandstream BT102 to my network. Its running new firmware (Ver 1.0.5.22 of 2005-01-21). I could NOT get the damn thing to (SIP) register.... Gripe 1: The New Firmware does NOT show the current version of all the firmware. You have to ask the phone manually with its menu button. Gripe 2: It does not show '****' in the the two password fields... This is what caught me - I