similar to: How to remote reboot Grandstream GXP-2000

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've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? TIA!!! Thanks,

Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

After working with the Grandstream GXP 2000 series phones, I have decided that I am quite unhappy with their problems, both voice quality, volume, features and others. For their price now, there are plenty of phones to choose from as well. So subjectively what would be the best Hardphone for a small/medium business with multiple line support, BLF, etc. Are the Cisco 7960 the best of the

Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and resumes the call. This shows that call was not dropped but phone lost connection with the server, whereas the caller on the other end was still talking. This is just unacceptable as this is

Hi, Aastra IP Phones have two configuration files on TFTP, aastra.cfg and <mac>.cfg. Both are in text format, which makes editing easy. And aastra.cfghas system wide settings and <mac>.cfg has settings for each indivifual phones. This makes it really easy to change the global parameters system wide by changing only one aastra.cfg file. On the other hand, as I could understand, for

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

For a while now I've been looking for a good web based soft phone solution, but so far no luck. A few solutions which I've tried, both Java based and Flash based, either don't work, or had bad sound quality. I need something which I could put on my productions server for my clients. Seems like good web based solutions are all paid ones, nobody is giving it for free. Any ideas,

Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice

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 all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some

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

Hi, I looked at the docs and probably missed it: is there a way to set a dialplan on the GXP-2000? (to avoid having to press "Send") Thanks, -- "Computers are useless. They can only give answers." - Pablo Picasso

Hi everybody, How can we add new contexts in asterisk realtime module? All I could figure out after googling is that a new context HAS to be declared in extensions.conf with 'switch => Realtime/@<databasetable>' under the context name declaration. This works fine as long as we are adding extensions only to this one context, but doesn't give the freedom to add new contexts for

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

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