similar to: Dlink DVG-1120 Linksys PAP2 any Good?

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

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

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.

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

So, I recently acquired an old DVG-1120M that was previously used with AT&T CallVantage service. It currently uses MGCP. I see online that there is a SIP version out there (DVG-1120S), which is essentially a DVG-1120M reflashed with the SIP firmware and the 'M' scraped off in favor of an 'S'. I've also seen people on this list saying that they flashed their M with S

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

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

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

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 list, I have search for a clear explanation about this mensage " WARNING T.30 ECM carrier not found", but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda

If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP.

I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David >>> asterisk-users@eol.ca 1/21/2004 6:39:34 AM >>> I'm not sure Primus

Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit

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

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

Hi, I have 3 PAP2 connected to *, they work fine but there are some things which I would like to improve, some of them are: - double ring tone when placing a call (I hear two tones it seems like the PAP2 is generating it's own tone) - some kind of noise (like glitches or something) when I pick up the phone (seems like some polarity thing) - I'd like to keep the tone after

The best I can do so far (which appears to be a bit of a hack) is (<:0>S0), which says to add a '0' to the start of the string and dial immediately. This gives asterisk an extension dialled of '0', which isn't the 's' that i'd hoped for, but is a good start! (S0) by itself doesn't work, nor does (<:>S0). Any other suggestions? Thanks James >