search for: dg104s

...I hear again those long+short beeps, but the extension dialed is not ringing. If I pres flash again I get back to the previous caller. On the console I see those digits that I dial after pressing flash. What is wrong? here is my mgcp.conf and exept from the console about this call. mgcp.conf: [dg104s] host = 172.50.0.1 threewaycalling = yes transfer = yes callwaiting = yes cancallforward=yes callreturn=yes context = test callgroup = 3 pickupgroup = 3 console: -- Executing Dial("MGCP/aaln/3@dg104s-1", "MGCP/aaln/4@dg104s||t") in new stack -- MGCP mgcp_request(aaln/...

I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

I am installing a used DG104S.... I got it to ring from gnophone, but all I got was fast busies. so I upgraded based on Pavel's link: ftp://ftp.dlink.ru/pub/VoIP/DG-104S/Firmware/MGCPDG104.zip So I now have: PROM Version: 3.0B22-D RUNTIME Version: 3.0B44-D But when I pick the phone up I get: ggdbg>000001...

It seems that the DG "gets lost" and keeps attempting to send RTP packs to asterisk and it get an icmp deny. The phones on that port will not work. Other phones do. So is this asterisk failing to hang up on the DG, or is DG not seeing a "call over" message? It is happening more frequently, but I am not at the location, so it is tricky to catch the streams live. Whenever I

...ss). On the flip side, I have run out of port forwarding slots on my LinkSys, so I am trying to resurrect my old startup's SonicWall PRO. They indicate they support SIP, but they do not elaborate upon that at all. Anyone have comments concerning the SW & SIP? So far I have an ATA-186, a DG104S, as well as a TDM400P and X100P with service from iConnectHere. I hope to ship either the ATA-186 or DG104S to Australia when I figure out how to deal with them remotely. Thanks, -- John

Hello, Anybody has it working with asterisk? Could you share your experience ( good/bad) Thank you -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

How do I do this 1) ZAP-> * -> IAX(1) ------> IAX(2) -----> DG104S ------> Handset 2) No Answer on Handset 3) Back to IAX(1) 4) IAX(1) tries a cell phone 5) Still no Answer 6) Local * Voicemail. I have 1 working, and I had 4 working when there was only one box, i.e. when the handset did not answer the DG, asterisk went to the next step. Now that I have step...

...ut can't negotiate a connection. I am using: exten => _95380024,1,Dial(Zap/1/${EXTEN:1},120,d) exten => _95380024,2,Congestion I don't have any problems doing a fax though my system. For this setup, I am running a simple Digium developer's kit on a P3-700 256MB. I have a DG104S and ATA-186 too, but I'm not using them for this. The ReplayTV box is connected directy to the TDM400P. I am running a relatively old build: Asterisk CVS-07/10/03-23:27:00 Any thoughts would be appricated. -- John (As a humorous aside, I just spent a few hours tonight trying to figure o...

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 messages,...

Hi, After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This

Hello I have a problem with my MGCP voice gateway. I use D-Link DG104S Boot PROM Version 3.0B38-D Firmware Version 3.0T86-D I tried asterisk v 0.7.2 and I am using latest CVS version now. When I dial a number very fast, or when I use a redial function, my asterisk receives coupled digits. My co-worker called number 245005111, these are a few lines of my debug. Th...

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have