similar to: using http to provision a Grandstrea GXP2000 phone

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,

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

Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is /srv/tftp. When config files are stored in /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra,

Using Samba 3.0.2 (specifically the samba-3.0.2-7.FC1 Fedora package) with LDAP as a passdb backend I'm encountering problems with Domain Groups. I have come across various postings, some to this list, with people that are experiencing similar problems. However I have not found any information as to the cause/solution. The problem is as follows. I have the following group configurations in

Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

Since 1.4.22 realtime status for sip peers seems to be broken. If I do a "sip show peers" from the CLI I get this: 2001/2001 192.168.2.234 D 5060 UNKNOWN Cached RT It is arbitrary which peers will say OK and which will say UNKNOWN and it changes over time. This is a problem with an application like the Asternic Flash panel because it uses the peer

Hi all, I've got a number of Grandstream phones and I'd like to be able to reboot them remotely, as I do my Polycoms... I've got this in my sip_notify.cfg: [grandstream-check-cfg] Event=>sys-control Doesn't seem to work. Any ideas? -- Take care and have fun, Mike Diehl.

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

We have several types of phones, cisco 7940/7960 aastra 55i/9113i/ grandstream gxp2010 I want to be able to implement hotdesking where an agent will logon to any phone. I got all of that working, without having to reboot phones, but then hit a brick wall. Voicemail. I still want each phone to use the BLF for voicemail indication, and to use the "voicemail button" to dial

Hello list. My phone rings, I pick up, and the conversation is terminated. Every time. The setup : Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server --> ITSP Could it be the SIP proxy of my Endian firewall ?? I have 4 accounts on the Grandstream which listen on port 5060 --> 5063. They have a proxy defined namely my Endian firewall. On this SIPproxy I have a

Hi, what's thereal use of cfg.txt file during Grandstream GXP2000 provisioning? Thanks -- Domenico Viggiani

Hi, Just wanted to let you know that we have just made our GPL toolkit "Gemeinschaft" available to the public. (Finally.) Mostly German for now - about half of the strings in the language strings file have been translated to English. I'm a software developer, not a marketing guy, so ... svn co https://svn.amooma.de/gemeinschaft/trunk gemeinschaft-trunk German readers: see

Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my

I am using 1.4, and have the above device, and it worked really well with monitoring 18 "hints" aka devices. Now, I've moved us to a hotdesking paradigm where the user is the "extension" not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ?

Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they

I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

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

I've tried to use a Cisco power injector to supply power over Ethernet to a GXP2000 phone without success. Although when I plugged these phone to a PoE capable Cisco Switch it worked without a problem! Knowing that all these three equipments implement IEEE 802.3af protocol, why doesn't it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho.

1. We just dial the extension directly and have speed dials setup for the first 6 parked positions. We don't use *8 at all. 2. Change the config on the phones under Account to "Send DTMF via RTP (RFC2833)" -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noc Phibee Sent: Thursday, February 08,

Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? Regards /ralf ________________________________________________ Ralf