similar to: GXP-2000 and transferring call directly to voicemail

I have a function *1 that starts and stops recording in a call. I use a function so I can use MixMonitor. It works well, however I would like to make it a little more integrated for my users. We have GXP 200 hardphones. So far I've been able to configure a softkey using the speeddial option to dial *1 during a call. I also have setup another key to monitor the status of recording (on/off)

>From the usual place, http://www.grandstream.com/BETATEST/GXP2000/ Note, there are two (and it took me a bit of a while to figure out) images to be loaded. Copy the ...a.bin's and the .bin's to your http provisioning directory, and reboot. The phone _must_ load the .bin files before it understands the ..a.bin files. After it loads the first one, the phone does lock up with the

Hi all Slightly OT but I know a lot of GS experts hang out here - I just upgraded a GXP-2000 to firmware 1.0.1.13 to try out the BLF functionality with Asterisk (which so far works as expected), but as a side-effect the phone won't sync with an NTP server - I've tried different server names (time.nist.gov and pool.ntp.org) and IPs in the config, but it refuses to update the time on

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

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.

Hello, I have some trouble with the BLF indicator. I have two phones that use the same hint: 13 => hint,1,SIP/phone13&SIP/phone13-wlan This works great from the asterisk side, but it seems the status change is too quick for the attached Grandstream-phones. When I ring the extension the hint changes to "Ringing". The Grandstream blinks. Great. Now, when someone picks up one of

Man, I really feel stupid, but after banging my head on a brick wall for several hours ... I need help! I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten 5707, and I've got an xlite on 5608. When I make a call from an outside line, I dial SIP/5608. The little blinky light on the GXP that's monitoring 5608 goes, well, "blink blink". :) I then press

Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). Thank you.

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,

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 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 a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before and all their old Lucent phones showed a button for each line. So, it was easy for anyone to say,

I've finally got a chance to play with 1.0.1.9, and the wiki has been updated. At the moment, I don't know of _any_ bugs with it. I'm yet to play with complex things like early dial, and will update the wiki as I find information. http://aussievoip.com.au/wiki-GXP-2000 I've just been given official permission to offer the firmware as a download, so if you visit

FYI, spoke with Grandstream this morning, the GXP-2000 release has been delayed again. Looking like April now before these hit the street. -- Cory Andrews Senior Partner VOIPSupply.com +++++++++++++ V: 800.398.VOIP X22 F: 716.630.1548 E: Cory@VOIPSupply.com

Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHz Box... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to

The vast majority of what I've done with Asterisk has been with the Grandstream GXP-2000's. These phones work great for us for everything *except* speaker quality is quite poor and appears to be half-duplex. So now that we've bought and are using 40 GXP-2000's we're doing some testing on other phones. I've bought a Polycom 301 & 501 as well as a Linksys SPA942. While

I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the

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?

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