similar to: Flashing DVG-1120M to DVG-1120S

I friend gave me his DVG-1120s after he realized that AT&Ts callVantage stuff would not work for him. It appears to be running a SIP version of firmware, however, it downloads an XML configuration file via SSL from AT&T. I cannot find a way to manually configure the VOIP portion of the unit via the GUI. I contacted D-Link to get an example configuration file so I could get it working

Hi, I have a dlink dvg-1120M (mgcp) box that i will like to use with asterisk. Is it possible? has anyone done that? Here's a link to the product page at dlink. http://support.dlink.com/products/view.asp?productid=DVG%2D1120M Also, does anyone has or know where to get the firmware for Dlink DVG-1120S (sip model)? thanks. -- Zahid

Thu, 06 May 2004 12:31:33 -0500 there was an e-mail to the list from Isaac McDonald suggesting to re-flash the DVG-1120M gateway with the DVG-1120S code to have the gateway login to *. I have been looking and can seem to find the flash code. D-Link tech support doesn't know much about the box. Any sources for the TFTP file would be appreciated. Once this is running SIP will I need the xml

I upgraded a DVG-1120M to a DVG-1120S. Everything works great, but I'm having some caller ID issues on incoming calls sent to the SIP device. Using debug on the device, the caller ID looks fine - just as I set it in Asterisk. However, the phone is showing "CID TRANSMISSION ERROR". Should I check the RX and TX gain levels? Try another phone? Any ideas would be much appreciated.

Hi, I am having problems with callerid name and the time with my dvg-1120S. Every time I receive a call, it reverts the phone to January 1st 12:00am. I've looked everywhere in the browser and telnet configuration to change this. Also, it never shows the name of the caller. I've even tried forcing the caller id info using SetCallerID, but it still doesn't show the name. Any

Hi, I am having problems with callerid name and the time with my dvg-1120S. Every time I receive a call, it reverts the phone to January 1st 12:00am. I've looked everywhere in the browser and telnet configuration to change this. Also, it never shows the name of the caller. I've even tried forcing the caller id info using SetCallerID, but it still doesn't show the name. Any

Hi Philip, Unfortunately, * speaks MGCP only as the Call Agent, rather than as the Media Gateway. MGCP is a master/slave protocol, and it would take some effort to make * work as the slave. I have the same problem: Free Telecom here in Paris includes MGCP service with their DSL. You can call any fixed phone in France at no charge! Rates to mobiles and international are quite aggressive, too.

I had just found a Dlink DVG-1120 on ebay and I'm curious if anyone has used you it with asterisk. They were only $65. I have tested with the Linksys Pap2 and found that box to be fairly nice except for a lot of backgound/white noise. I was wondering if any else had experienced that? Let me know if I've wasted $65 on the Dlink and also if you had similar experience with white noise on

I am trying to connect directly to AT&T VoIP service CallVanage. I have ATT's ATA (D-Link DVG-1120M). They use mgcp. I have traces of the connects from the Dlink and hoping to setup Asterisk the same. It looks like I need to have Asterisk be a MGCP endpoint (gateway). How do I configure this? Does the mgcp.conf support "register" like sip etc? What is the syntax?

Hi all, I want to link three incoming Bell Canada centrex pstn lines (which currently go to an old norstar pbx) into asterisk. Can anyone suggest the most "painless" (i.e., "just works") way to do this? Has anyone used the D-link DVG-3004S four-port FXO-to-sip adapter, or the twice-as-costly Audiocode MP104-FXO-C3S? I know the Digium TDM400P or Sangoma A200 are options

We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only

Hello, I get warnings in my asterisk log: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call. I've used sip debugging to figure out the cause. It's my D-link DVG-1120S that don't understand message-summary events that asterisk sends out for MWI indication to the client. Is there any way to disable this in asterisk for this particular client? Tanks in advance, Magnus

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OpenSSH Unix Dev, Mac OS X 10.5 recently shipped with OpenSSH 4.5p1. This build includes a number of patches, some general bug fixes and some platform- specific fixes and enhancements. These patches are available from our open source site (http://www.opensource.apple.com/darwinsource/10.5/OpenSSH-87/ ). Following is a brief description of each patch. We'd be more than happy to

Does anyone have any experience using this DLink quad FXO <> SIP gateway with Asterisk? I'm still looking for an analog interface that I can live with, having tried X101p, SPA-3000 and TDM400...all with less than desirable results. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power

I am interested. I'm using Callvantage and Centillium TA. I'm also using Asterisk with a X100P Card. Eventually I want to replace this with my Linksys NSLU2. I'm having mystery power issues with my linux server. Currently I have calls from soft and ip phones going to asterisk then out to the ta fxo to fxs. Regular pots phones go to a telephone distribution panel with the

On Wed, Nov 11, 2020 at 5:47 PM Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: > I?m sure you can reflash LSI card to make it SATA or SAS HBA, or MegaRAD > hardware RAID adapter. Is far as I recollect it is the same electronics > board. I reflashed a couple of HBAs to make them MegaRAID boards. > you can reflash SOME megaraid cards to put them in IT 'hba' mode,

> On Nov 11, 2020, at 5:38 PM, Warren Young <warren at etr-usa.com> wrote: > > On Nov 11, 2020, at 2:01 PM, hw <hw at gc-24.de> wrote: >> >> I have yet to see software RAID that doesn't kill the performance. > > When was the last time you tried it? > > Why would you expect that a modern 8-core Intel CPU would impede I/O in any measureable way as

I have a issue with two Zap clone cards where they used to work. I am using Asterisk@Home 2.5 which includes Asterisk 1.24 and Zaptel drivers 1.2.4. The system is a new Intel Celerion machine. I used to have the same cards running in a Intel PIII system. in this system, they worked. In this older system, I was able to call into the machine and call out from it. Now that I have upgraded

Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20 Cdn a month!! Local number portability!! Cheapo Primus LD rates!! They don't care where geographically you plug it in!! When you sign