similar to: VoIP GSM Gateways

I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061028/87014bd9/attachment.htm

The long waited Ultimate GSM Gateway is finally out. This time we have managed to source a new patch of brand NEW GSM Gateway at prices that is only 50% of what the market rate. And with the SMS Function and many more... For purchase please email gsm AT cyper-telecom.net. We accept paypal and bank transfer. Postage is not included. Please notice we have also got the standard Dual Band GSM

I've been asked to purchase a gsm gateway for use with our asterisk server (for our use, not reselling) I have a spare ISDN port on the server, so I have use either a PRI or VOIP gsm gateway. What would people recommend ? Has anyone used the QuesCom 400 ? I would also love to know a rough idea of cost ;) Once I've gotten the info, I'll post a message on the biz list for a

Hi, I use a simple script to query a 2 table's in a MySQL database. This is called using AGI. The first one has got about 200 rows in it each one is a caller ID number and a name. If I get a match then I set CLI to "name <number>" The second table has over 10,000 rows in it. It has all the telephone exchanges in the UK in it with the prefixes for the exchange. If I

Has anyone used the Swissvoice IP 10S (www.swissvoice.net) VoIP Phone with *? Adrian -- Adrian Walker adrian@digitaltraffic.co.uk ======================================================================= This email has been scanned for Virus infection by MessageLabs For more information please contact messagelabs@atomwide.com

More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30

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Selling analog GSM gateways at 119.00 USD. Polarity reverse for "call answer" and "call end". Compatible with all Digium/Asterisk analog FXO cards. Suitable for low-traffic call termination, inbound calls from cellular network or for phone bill optimization applications. Main Functions - Provide Accurate Anti-Polarity Signal - Incoming Calls Display - Add Prefixal

Hello everybody, I have an Asterisk box with a TDM04B and would like to connect it to a GSM Gateway. Can someone tell me whether i can get the callerid for incoming calls in this case? Thanx Lamine

Hi All, Has anyone on-list tried using a USB style adapter like the VTA-1000 to provide some form of gateway from Skype into *? If so, how well did it work? Would I need to combine the VTA-1000, which appears to be USB-to-RJ11 and a standard FXO port? Would it be possible to provide a more direct approach? Some of my associates use Skype, which I'd prefer to avoid. However a single

I am impressed, I have been trying this for sometime using the SIP image and the only difference I can create is a 'single' and a 'double' ring on the phone. I use the 'single' ring for phone calls and the 'double' ring for the doorbell. I would love to be able to choose a ring tone based on the incoming msn or callerID. The idea of the phone shouting 'Its the

The feedback we are getting so far has been excellent! As more is decided the list will be updated, if you'd like to be involved in helping, please join us on the IRC channel, #asterisk-uk on irc.freenode.net. If your company would like more involvement with the event, please email me directly. I would really like to hear from people/companies who would like to: - # Exhibit a product or

Hi, there. I have asterisknow beta 5 with the following data: Ip 192.168.0.60 mask 255.255.255.0 gw 192.168.0.1 the router (a linksys) has port forwarded the port udp 5060 and from 16384 to 16482 udp-tcp from the internet to the asterisk machine. the only protocol allowed is g729. Which work fine for the ip phones I already have setup in the LAN. My problem is trying to register to a voip

I have recently upgraded my firmware from v6.3 to v7.4. Now when the phone is booted or rebooted, the initial screen "Initializing Vlan" takes forever to initialize before it initializes IP. Any ideas/Thoughts? (Trying not to Revert back to v6.3). -C -------------- next part -------------- An HTML attachment was scrubbed... URL:

I bought a MV-372 for 2 SIM cards as the one channel model seems to work well (see http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk). The setup is such: --------- Inet --> VoIP provider ---> POTS | | (iax2, NAT) | asterisk (on abox with iptables fw) | (SIP, LAN) |----------> SNOM190 phones | ----------> SIP-GSM-module ---> SIM

Hi, Does anyone know if it's possible to have 1 port of TE406P to provide T1 PRI to other VoIP gateways? I am trying to provide T1 PRI to one of our old Clarent gateways to integrate it to Asterisk. Can anyone help? Thank you. Andy

I am interested in the wellgate 38XX FXO and FXS gateways (and other similiar units). My question is can outgoing call files use these devices??? Can I fashion an outgoing call file with a channel like: SIP/WellGate-1/5551212 (for the first port) or SIP/WellGate-2/5551223 (for the second port) TO these devices behave this way? Of course incoming calls I dont see as a problem. It's asking

Hi all What headset do people use with the GXP-2000? Any recommondations for or against particular models? Thanks Peter -- Peter Bowyer Email: peter@bowyer.org Tel: +44 1296 768003 VoIP: sip:peter@bowyer.org

I was hoping someone might have some knowledge to impart regarding VOIP PRI Gateways or the psudo ISDN services being offered these days. The official line in Australia is that "true" ISDN services are on their way out. I am testing a service provided by one of the telcos I am told that it cannot provide ISDN cause codes for disconnected/invalid numbers and all we get is the audio. Also

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark