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Dear all, Just wanted to know if any one had deployed the Grandstream GXW 4024 yet. Wanted to hear any feedback and/or problems with this unit that you may have experienced. Thank you. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz

I have an Asterisk running with both GXW4008 (FXS) and GXW4108 (FXO). The FXS Gateway works perfectly, no problem so far. The FXO Gateway (GXW4108) also works fine. The configuration for local settings in Brazil was quite easy, however, I still not able to make Caller ID to work. I'm setting as DTMF Caller ID type, but still not work...

I'm trying to use a GXW-4008 for the first time to provide simple POTS. Is anyone using it? How about samples of SIP.CONF and EXTENSIONS.CONF? Do you have advice for configuring the GXW-400x for this application? How long a local loop will it support on the FXS ports? When I started to configure the unit, I...

Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesn't work. Someone has a good configuration for this gateway? Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20...

How well does the Grandstream GXW-4104 or (8) work w/Asterisk? I would use a Cisco Switch w/FXO Ports but that would be a little "Pricy" I Can't use a Digium FXO Card, as the asterisk Server is offsite. Thanks, William Stillwell KI4SWY ________________________________________________________________ Sent via t...

Thought i would share this so it doesnt annoy others as much as it did me :) If you recently installed a GXW 40XX and your extensions start ringing magically now (ringing for no reason, pick it up its a clear tone) you need to check the "Disable send MWI" in your gateway. apparently certain old phones do not like the MWI signal and treat it like a ring tone. -- Faraz R Khan Chief Arc...

...currently looking to deploy an Asterisk server using an FXO media gateway to connect to the PSTN and was looking for any user experiences that may aid in selecting a gateway. Specifically i'm looking for a 4-port model under 500 dollars. Within this category exists: MediaTrix 1204 Grandstream GXW-4104 AudioCodes MP114 I've read over Voip-info.org regarding these products and numerous mentions of problematic setup and configuration seems to be a common theme. Can anyone provide a recommendation based on user experience? Feel free to suggest an alternative gateway if one stands out. Th...

Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issue...

Hello list ! I don't have the money to test out all the products and reading the manuals is not always that enlightening... Does someone here know a good gateway-product that lets analogue telephones communicate with an Asterisk-server. I have found the Grandstream GXW-400x to be able to add SIP-accounts to analogue telephone devices that are connected to the FXS-ports. Moreover this product has a backup-PSTN line for emergency calls and backup. Could you advice other products/manufacturers ? Greetingz, Jonas. -------------- next part -------------- An HTML att...

Hi, I have an 8-port Grandstream GXW-4008 V1.2A ATA converter with analog phones connected to it. They work fine except for just one "feature" I would like to modify. Somehow, each time the ATA re-registers the SIP clients or each time the device has to be rebooted for maintenance, the phones ring once. This feature can be...

Hi, I am trying to fix a Asterisk setup with buggy (POTS) Fax machines. The setup consists of the following components: - A Digium TE121 for connectiong to E1 ISDN - Debian box with Asterisk 1.4 - Grandstream GXW-4008 SIP ATA to which the Fax machines connect I am aware of the problems with this type of ISDN <-> Asterisk <-> SIP ATA <-> Fax machine installations, e.G. different "clock source" of ISDN and the ATA, ethernet timing issues, etc. So it's no suprise that i encount...

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

Hello I have a working * server with a TDM card and 4 FXO ports. We have 4 lines now and need to add 2 more lines (and possibly two more later). I'm wondering the best upgrade path for this situation. The simplest I can invision is adding another TDM400 card with 4 FXO ports, and use 2 now and the remaining 2 later. Are there success stories with using 2 TDM cards? Any info will be

Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP trunks, and a second (PBX) which has UAs registered to it. We have a customer that has an existing pbx that we trunk analog lines to using a GXW-4008. The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF. The issue I'm currently having is with inbound DTMF. PBX and PSTN are connected through a standard sip trunk. Both machines are on the same physical switch. Here are the results I've seen: PBX <-&gt...

I own a TDM2400 board, with three FXO modules and one FXS. I'am having trouble with analog sip phones, from two different equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), sometimes when I am calling someone, then I press flash, and then call someone else, both calls stay connected after I hang up. [Sep 29 07:18:06] VERBOSE[3218] logger.c: -- Called g2/16 [Sep 29 07:18:06] DEBUG[3218] chan_dahdi.c: Sent deferred digit...

I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone "unconfigurable" via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the "keypad update" feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited and I can...

...am 4 FXS Gateway and my *XXXX extensions are not working. I disable the special features and changed the DIAL to {X*#+} but not luck. I can dial any other number, receive calls and so on. This is the only thing that seems to be an issue. Has anybody found a way around this problem. Reference: GXW-400x IP Analog Gateway Series Documentation: http://www.grandstream.com/gxw400x.html -- Yours truly, Yu

I have a FreePBX system with PRI trunks that's doing a number of things very nicely, but frustrating me in one area. I am using a Grandstream GXW-4008 in an off-premises location to provide "POTS" service on four ports (this device worked fine in an early application using a hardware VPN to the Asterisk server). The Grandstream has a public static IP port, as does the Asterisk server. Extensions 1021, 1022, 1023, and 1024 r...

...hours research and here's what I was thinking: FAX A <-> GS-HT286 <-> |----------| | | |--------| <-> POTS A FAX B <-> GS-HT286 <-> | Asterisk | | GS | <-> POTS B | 1.4.2 | <-> |GXW-4104| <-> POTS C FAX C <-> GS-HT286 <-> | to route | |--------| <-> POTS D | calls | FAX D <-> GS-HT286 <-> |----------| What I need is if FAX A calls out it goes over POTS A and if POTS A rings FAX A rings, etc. I'm confiden...

Hello all, i read a few articles online about the possibility to setup a "buzzer" door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! _________________________________________________________________ Connect to the next generation of MSN Messenger?