similar to: GXW4024

Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign

Hi, In this http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/204725/focus=221375dating from 2008, experiences with Grandstream GXW4024 were asked. Has anyone something up-to-date to share about this ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091005/ad69fa22/attachment.htm

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, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue

Hi All I've done some reading on the wiki and read some of the mailing list archives, but can't see anything on this. I guess this means I'm either searching on the wrong thing, or have totally the wrong idea... Can anyone suggest if the following is possible? Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have

I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are

Hi, In our design of Icecast2, our multiple sources will stream their high quality audio via a dedicated audio encoder (not a computer): Live audio > L+R microphone > pre amplifier > audio encoder > Icecast2 VPS So far I have a shortlist of 4 possible brands of audio encoders: Sonifex PS-SEND, Barix Instreamer, Bric-Link, Outcaster OC100. All have their pro's and cons.

Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has

I've tried to figure out from the web site the minimum hardware cost to run a small office Asterix solution but I'm afraid I miss something: Let's say that I want to connect four/five analogic extension to the PBX. I have: - 1 computer as the server (with linux and Asterisk on it) - 1 dummy patch panel to connect all the analogic phones around the office What (and how many) cards

Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb

I am looking to build a small PBX for an office that has 3 incoming analog lines and less than 10 extensions. For the Asterisk server I am going to use a small form factor PC with no-PCI slots so the FXO interface needs to be either FXO->SIP or USB. Can anyone make suggestions? I am looking at an AudioCodes MP114 FXO or possibly two Sangoma U100's but don't have experience with

Dear List, I'm having considerable trouble setting up my environment (Linux, Fedora 16, Bash) to build and check packages under R Under Development (r61660). I'm doing this to better get a handle on difference in the output from running checks on examples in one of my packages. Note I compiled R Under Development myself The problem I am now having is whenever I try to build my analogue

Hi Thomas, As per your question "Out of curiosity do any of those support anything else than MP3?" - - > answer is yes. See attached link an overview of 6 hardware audio encoders (Link to pmg file via WeTransfer http://we.tl/w0H6HJsxLd ). All offer various output formats, not all matching ICECAST input requirements. Not all offer an authentication on mount level. Price

Over a year ago I saw a discussion about a standalone device which converted a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device is? (I'm looking for a standalone device - not a PCI card). Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070907/7060b23e/attachment.htm

Hello, I bought a Grandstream Handytone 486 to forward incoming calls from our old analogue PBX to the asterisk server. My first test was connecting an analogue phone to the Handytone and calling a sip phone - worked. Now I used the same cable to connect the line port of the Handytone to the analogue pbx. When I call the number of the analogue PBX I hear a clicking inside, but the call

I have the following data: prefix <- c("cheap", "budget") roots <- c("car insurance", "auto insurance") suffix <- c("quote", "quotes") prefix2 <- c("cheap", "budget") roots2 <- c("car insurance", "auto insurance") roots3 <- c("car insurance", "auto

hy all actually i want to have a setup of five offices having round about 200 extensions ( each office having 35 to 45 ) which will be connected through asterisk. now either i should go for voip phones( hard phones ). or use any interface card to asterisk server to which the analogue phones will be connected. ----- if i use analogue phones in the above case ( we have analogue phones already )

I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a

Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the

So made my first forray into 1.4 and DAHDI and hit a problem. (Not convinced this is a DAHDI issue though...) Testing an analogue line and asterisk sees the caller ID being passed, but then fails to detect ringing. A plain old analogue phone plugged in rings just fine. Console output: == Starting post polarity CID detection on channel 4 -- Starting simple switch on