similar to: DVG-1120S no call display name and time

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

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

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

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.

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 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

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

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

We have extra equipment that was over-ordered or unused. All of the equipment is brand new. The equipment has been highly discounted to move quickly - the last set of equipment sold in 48 hours. If this equipment is of interest to you, call or e-mail quickly. Buy on VOXILLA and SAVE $300 each (Cisco routers & switches): http://store.voxilla.com/customer/home.php?cat=259 For Sale (all new):

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

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

Hello !!! I'm trying to setting up my DVG-2032S Voip Gateway to use Asterisk as outbound proxy, that's because I already have this gateway before to begin to play with Asterisk. Every time when I enable the OutBound Proxy option and call from my Ericsson PBX I got the follow message in DVG-2032S System Information: Hook off, and nothing in Asterisk log or in the console. I've

Hello How do I fix this.... [IPComms-in] exten => ${IPCCIDN01},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN01},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN02},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN02},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN03},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN03},2,SetCallerID(${CALLERID}) exten =>

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

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.

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

I'm trying to connect an asterisk server via oh323 to a Lucent iMerge. I patched the code due so that Lucent can handle asterisk's ver4 h323 http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration I can now successfully dial in to our company over multiple lines. The issue is when I dial out The first outgoing call connects to an outside user A The second call drops the first

I am trying to SetCallerID to a variable I have defined. This obviously is wrong. It actually sets the caller ID to $NEWCALLERID. I have search through the examples on wiki but wasn't able to find something similar to see what I was doing wrong. Could someone tell me the correct way to SetCallerID to a defined variable? exten => 2125551212,5,SetCallerID({$NEWCALLERID}) exten =>