similar to: Grandstream GXP 2010 : multiple accounts not working

[sorry for my maybe newbie Qs and using this group for my busybox issue but though having visited some forums about this I'm getting nuts] I unslugged a Linksys NSLU2 device with V2.3R63-uNSLUng-6.8-beta as I plan to use it as a DC replacement for my W2k server at home. But I already got stuck before this. Following the unslinging I installed the Samba3 IPKGs for the NSLU guided by the hints

When I mount a drive via Windows XP and copy a file via Samba, the file appears to be copied. When I refresh the window in Windows Explorer, however, it disappears! When I copy the *same* file a second time, Windows asks me if I want to *replace* the file (as if it were there but I just can't see it). I've attached the Samba drive to my Windows XP machine and I can see the file that I

Hello there ! I have successfully installed Asterisk on a normal server and on a Linksys WRT610N with DD-WRT running from connected USB-stick. This is not my first Asterisk-installation. I'm always finding a smaller and smaller platform to install Asterisk on. But now I encounter a problem. I'm using the same Grandstream GXP2010 with a of my installations. Every time registration goes

Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register => 092779077:XXXX at 85.119.188.3 This the output of SIP show peers : asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status

Hi All I have a Linksys NSLU2 device which is used to hook USB2 drives upto my network as network attached storage. The Linksys firmware upgrade for this device includes samba 3.0.11 which is a non-starter regarding OS/2 connectivity. There is an alternative firmware based on the Linksys firmware called Unslung from http://www.nslu2-linux.org/ The Unslung firmware allows

Hello list ! SETUP : Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk (VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone PROBLEM : I've noticed that when I put down the horn of my Grandstream it still takes a while for my GSM/CellPhone to stop ringing. INFORMATION : This is the output on the CLI of the local Asterisk-server : [Oct 3 17:40:33]

asterisk*CLI> sip show domains Our local SIP domains: Context Set by jocan.local (default) [Configured] 192.168.1. (default) [Configured] [Jun 26 17:49:03] NOTICE[5570]: chan_sip.c:15889 handle_request_register: Registration from '<sip:grandstream at 192.168.1.248>' failed

Hi, I have a writable and readable file on a small network file system (Cisco NSLU2 Unslung; non-NTFS) that I access via a mounted drive on Windows Vista. My problem could be due to a "funny" file system/server, but here it goes: > pathname <- "Q:/foo.txt" > cat(file=pathname, "Hello world!\n") > readLines(pathname) [1] "Hello world!" >

I have installed gnutls and gnutls-devel from RedHat repositories [root at asterisk asterisk]# yum install gnutls gnutls-devel I have installed iksemel with gnutls support : [root at asterisk asterisk]# cd /usr/src/iksemel-1.3/ [root at asterisk asterisk]# ./configure --with-gnutls --prefix=/usr [root at asterisk asterisk]# make [root at asterisk asterisk]# make check [root at asterisk

Guys. I just downloaded the recent firmware for GS GXP 2000 and I must say the phone works great but... How do you make the conf button work?????? Anybody done that? Also, with great dissapointment I must ask, where is ILBC support? GS web page mentions it and the manual says it supports it almost using bolds :) soooooo where is it???? Any light on this?

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.

Is anyone else having an issue with GXP-2000s and transmit gain? All my other phones are fine on my TDM400P with txgain set at 0, but the GXP-2000 caps at about a third of the scale in ztmonitor. I'm getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 320 and Polycom 301 are great, and my Budgetones are overmodulating. Is there any conceivable fix on the

Hi, I have a lot of GXP-2000 phones not registering with Asterisk server. After two days of attempts, it seems that problem is due to the fact that phones and server are not on the sme network. Do you know if this is known issue? -- Domenico Viggiani

I have about 30 GXP-2000 phones running 1.0.1.9 which have all been configured using the provisioning feature so the configuration is all identical. The problem I am having is that they randomly seem to stop registering with asterisk. When they stop registering they can still make calls but oviously asterisk cannot ring the phone so all incoming calls go to voicemail. Has anyone else had similar

I started out a few years ago with some SPA-841 sets, because the Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more call appearances, and I didn't want just the 4 max that the SPA offered. As it turns out, with the greater flexibility of VOIP, I don't need 'dedicated' CAs the way I needed them on ISDN previously, so 4 is actually adequate. Along the line,

Anyone used the new Grandstream GXP-21XX series phones. We have been testing these phones and like what we see. We are looking for a greater cross section of testing before we roll them to production. Any feed back would be appreciated. We are talking with Grandstream engineering and they are looking for feed back as well. Any input is appreciated. Thanks Bryant -------------- next part

Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus

Does anyone know the secret to get the GXP-2000 Message waiting lamp to illuminate? Or can point me toward some docs that might explain it? Thanks! --Jeffrey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060501/7ffc668f/attachment.htm

Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use the SPA-941 the call terminates on the other right away soon as I hang up. I have updated the

I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13. Is anyone else getting the following error in the asterisk console: [Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short every couple of seconds when a handset is in a call? I didn't notice this happening when I was using an older GXP2000 with the same firmware (doesn't mean that it