similar to: custom configuration with appliance aa50.

Hi All, Thanks for all of the help I've been given in the past and info I've picked up from this list over the years. I have an "official" Asterisk appliance (the AA50) running my PBX at home (we previously also had an AA50 in a satellite office-that one was recently retired and replaced with Asterisk running on commodity server hardware). Anyway - the AA50

Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there, I am looking to buy an Asterisk Appliance or Box for my organization and was hoping to ask for recommendations. My ideal box is a small device in size like Digium's AA50 Asterisk Appliance ( http://www.digium.com/en/products/appliance/ ) but will still have these technical features : -> Run Linux obviously -> Run a fullly configurable distro of Asterisk (not embedded) -> Can

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 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

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

Hi, I have several GXP2000 phones which used to work fine with Asterisk 1.2. However, after upgrading to Asterisk 1.4.21.2, whenever I initiate a call from a GXP2000, it gets dropped after 20 seconds exactly. I have "early dial" enabled on the GXP2000 and "pedantic=yes" on the server. If I disable "early dial", all works well ("early dial" or "overlap

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

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

From: "Richard W.M. Jones" <rjones at redhat.com> This allows custom parameters to modify parameters added by libguestfs, eg. by doing: -set drive.hd0.file=rbd:foo/bar Thanks: infernix @ #libguestfs --- src/launch-appliance.c | 31 +++++++++++++++++-------------- 1 file changed, 17 insertions(+), 14 deletions(-) diff --git a/src/launch-appliance.c b/src/launch-appliance.c

I've been scratching my head for a while and I expect it is my mediocre knowledge of Asterisk which is holding me back. If anyone can assist me with some pointers I'd be grateful. Basically, I've hooked up a Viking intercom at the front door. It hooks into an fxs as a "phone". Up till now I've just played back a "go away" message if any internal phones are

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,

Does anyone know how much space the appliance has for voicemail and/or logs? Doesn't have an embedded disk from what I can see, and only a 1G flash card? -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines and for the most part everything works. Dialing out on 23 lines to phones works fine. I have to use the Local channel to call the intercom system (from call files). If I only call 1 intercom system at a time so it uses DAHDI/1 everything seems to work as I can call all 8 intercom systems and play a message. The

Hello all, I'm having a problem with my intercom function that I use for under-chin paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's for our general phones. I have a global defined which has all the SIP channels concatenated together - this is ${ALL-PAGE-EXTS}. The problem comes when a user is on the line, and someone else uses the intercom function to page

So, after playing with 6.0 on the Cisco 7960 and 7940 platforms, I have the following gripes, which I've sent to a very clueful Cisco person already. Mind you, I love the Cisco 79xx series phones, and currently they are what I recommend to anyone who wants a 'real' IP phone. I just cringe - Speed dials. It's nice to now have speed dials in the line appearances that

Environment: Asterisk 1.2.14, FreePBX 2.2.0, Aastra 480i IP telephones firmware version 1.4.1.1077. Problem: Intercom feature: the dialed phone does not play the splash tone when auto-answering an intercom call. Otherwise, intercom works perfectly. Questions: What is the extensions.conf syntax to trigger a splash tone in Asterisk 1.2.14 (from the documentation and posts I've found, it has

Hi. I'm currently setting up a particular conference: 3 members (a,b,c), a can speak with b and c, b and c can speak only with a and not between them. I found my possible solution with paging/intercom using option "d" (full-duplex), but I need to make ringing the phone in intercom. Now I set auto-answer=6 but after first ring the phone hangup the call. There is a way to using

Hi, I've had a bit of a problem with one way audio, and it happens exactly when I believe it shouldn't (and works perfectly when I would guess I could have issues. Setup: GrandStream GXP2000-------Linksys Router-----------Internet------Asterisk box (hosted somewhere, fixed IP, no NAT) ----------- VoIP provider -------PSTN When a call comes in from the PSTN, the call goes all the way

A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP