similar to: Grandstream FXS Gateway star codes

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

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'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 was able to connect via the WAN port. Now I'm unable to connect to

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 the WebMail system at kotbh.net

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 issues with using

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.

I'm 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

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 working. Let us know what kind of problem you have, maybe I

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

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 useful as it notifies the user of the re-registration.

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:

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

Hi all and thanks for reading. I am experiencing a frustrating issue with asterisk where on some calls the volume suddenly drops to inaudible o completely fades away for a time. If you hold on long enough (20 to 30 seconds) the sound will come back. My asterisk server is on a public IP, and basically acts as a VoIP bridge receiving calls from my customers (all of whom use Grandstream GXW400X

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

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

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

Hi Guys.. Just wondering if someone could help me with a few questions please. were currently using the ulaw codec with our grandstream/iconnect/asterisk setup and its working pretty good except for the fact it downloads heaps. Does anyone know a good site to get referances to how much each codec downloads/quality etc etc ? Ive tried using that g723 codec but i have have problems as soon as a i

Hi, Does anybody know how to properly execute a transfer (without using the |Tt option) from a GS100? Scenario: 1. I call from X-PRO (ext 1100) to Grandstream (1101). 2. Grandstream answers. Call is established. 3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold. Grandstream gets dial tone. 4. Grandstream dials 1103 (the extension of another GS100). 5. Grandstream hangs

Hi, Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? -- Best Regards, Miroslav Nachev

Voicemail works fine for me. Post console output here to let us know what went wrong. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Breeden Sent: Friday, March 25, 2005 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: CAUTION: Re: [Asterisk-Users] grandstream