similar to: ATT CallVantage & Asterisk

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.

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

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

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

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

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

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 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

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

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

Hello, I try to place a call between CISCO IP phone 7905 (h323) and an analogue telephone on DLINK DG-104S (mgcp). Call setup is handled ok - both way, but after connection of the call I get both site no-audio, even codec are Set to alow-64 both. The call is terminated with this debug output - Didn't get a frame from channel: MGCP/aaln/1@dlink-1 and is Hanged up. Jan 18 20:32:55

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

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'm playing around with Asterisk and DPH-100M (Dlink mgcp phone) on my debian box. I've got stable version of Asterisk (packaged for debian) working with dlink phone and 7910 from cisco (minimalistic extensions.conf and chan_skinny for 7910) Everything works fine. Now I'm trying to get CVS HEAD working with MGCP. I want to test chan_sccp

Hello, I've successfully installed Asterisk from last CVS and configured it for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip server. All are work fine at G711 codecs, but then I disable all codecs except g729 some calls failed (Not all calls. Some calls passed at g729 succesfully). All my devices configred to use only g729 and I don't see other codecs at mgcp or sip

Hello, I`m try to make the attended transfer work Dlink DG-104S via FLASH, when somebody calls my phone I pickup and press flash to get a second line to call another extension. When I press flash I hear no dialtone, and only a long and then small beep. When I try to dial digits I hear again those long+short beeps, but the extension dialed is not ringing. If I pres flash again I get back 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

Hello Im having strange asterisk ring. I'm dialing PSTN network, then I get my call answered and I hear a person talking but the same time remote person can't hear me. They get a ring tone. What can be the problem? Where do I need to look for it? have no clue. Running Asterisk 1.2.14 svn rev 48468 Voip gateway is Cisco5300 with IOS 12.3(9) Scheme: 192.168.1.201