similar to: setting a dialplan on a GXP-2000 Grandstream

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I

Hello, At a client site yesterday I installed a dozen GrandStream GXP-2000's with 1.1.0.13 firmware but I had to backtrack and reactivate the old PBX and phones: network access for users windoze PC's through the phone's switch port was unbearably slow, making it almost impossible to work. When plugging back PC's directly to the LAN speed was normal again. On my test setup

Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc.

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Hi, I have 2 linux accounts on different machines (same login, same password). Can you please tell me how I use rsync directories between 2 accounts? 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,

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

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

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

Hi all What headset do people use with the GXP-2000? Any recommondations for or against particular models? Thanks Peter -- Peter Bowyer Email: peter@bowyer.org Tel: +44 1296 768003 VoIP: sip:peter@bowyer.org

Can anyone here have experience using G.722 on the Grandstream GXP-1200? It's supposed to support the codec, but I wonder if the handset does it justice? The older BT-200 also supported the codec, but the handset was not good enough. You could only hear the improved call quality using a headset. Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at

Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Cheers, Richard. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060109/26c4a63c/attachment.htm

Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was

I have upgraded the GXP-2000 to the newest firmware 1.0.1.12<http://1.0.1.12>and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again

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.

On my Grandstream GXP 2010 I have the possibility for 6 channels and thus 6 different accounts... Line 1 I define an account that registers directly to an online Asterisk-server, somewhere in a datacentre. Line 2 I define an account that registers to the local Asterisk-server (NSLU2 unslung) When I activate both accounts, only the first account (to the Asterisk-server on the internet) registers.

Hi, I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by "zap show channels". I tried adding "dahdichanname = no" to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel

Hi, This used to work fine in 1.4: exten => 2131/,1,NoOp(reject3: ${CALLERID(num)}) exten => 2131/,n,Playback(no_unknow_callerid_here) exten => 2131/,n,Hangup And now, after upgrading to 1.6.1.x it matches every callerid. Did something change? Thanks,

Hi, Is there a way to have markdown automatically convert obvious (http, mailto) URL's to links? i.e: http://example.com -> <a href="http://example.com>http://example.com</a> Thanks, -- http://www.critikart.net