similar to: Rebooting a Grandstream

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

Hi, Can you give me some councils of remotely rebooting sip phones in asterisk server? How to configure sip_notify.conf and sip.conf? Kind regards, Guan ; Reboot Polycom Phone Event=>check-sync Content-Length=>0 ; Untested (Reboot Sipura Phone) Event=>resync Content-Length=>0 ; Untested (Reboot GrandStream Phone) Event=>sys-control ; Untested (Reboot Cisco Phone)

With Asterisk 1.4 I could use commands like: /usr/sbin/asterisk -rx "sip notify reboot-snom mjc_home" to reboot a snom phone. Now, with 1.6, when I try that, I get: Unable to find notify type 'reboot-snom' Command 'sip notify reboot-snom mjc_home' failed. Do I need to add some magic to sip_notify.conf? I haven't quite figured out how to make it work. - Mike

I have a GXP2010 phone, the one with 18 blinky lights ;) I currently provision the phone by writing out the conf file, encoding it and sending it to the tftp server. I was wondering if anyone had managed to automate the web side of provisioning ? TIA Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

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

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Hi all, I've got a new server up, but it's not staying up.... After a day or so, it segfaults with: [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.2(a) Driver]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use

Hi all, I'm trying to configure a few conference bridges. I've started with the very basic: [general] [default_bridge] type=bridge [default_user] type=user [default_bridge] type=bridge [5340] type=bridge However: confbridge list Conference Bridge Name Users Marked Locked? ================================ ====== ====== ======== *CLI> It doesn't seem to be

Guys I just read on the wiki: "2005-07-19 - long awaited extension lights (hint priority) and call pickup on various phones work with newly released asterisk patch digium bugtracker - feel free to test and report findings to the bugtracker to have this commited to cvs." How does this work? And will it work only on certain phones or can it work with the gxp2000?

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great

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

Hi all, I've got a device that seems to become unreachable for about 2 minutes, every hour. From what I can tell, it isn't due to network or server issues. Any ideas? TIA. -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the features.conf file, you can view it at: http://www.diehlnet.com/features.conf Any ideas are welcome. TIA, -- Mike Diehl

Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other

One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on .... not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Thanks! Kind Regards, Erik

OK, now this should be really simple, but I am a bit of a newbie so please bear with me. I have an *@Home box setup with TDM04B and two POTS lines. On the SIP side, I have GXP2000 phones. Most things seem to work, but the users cannot figure out how to transfer incoming calls from one extension to another. Now I am not sure that I have things setup correctly, but is there something