similar to: DG104S firmware has error?

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.

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

Hi, Is there a way to use swissvoice IP10 in MGCP mode behind NAT. I already use this fonctionnality with a PBX other then asterisk, and it work very well. In fact, I need to use this settings in the IP10 telnet session set xgcp rgw_name MAC_ADDR set tcid 0 notify_entity ca@[@IP PBX]:2427 In fact, the phone register with the MAC@ and not with the IP@, and the PBX reply the phone in the UDP or

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. The identifier of first digit

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

I've searched the archives a bit and have not really come up with a good answer to my queries. I have * running on a RH9 box behind a LinkSys NAT box. I can talk with iConnectHere outbound just fine. I am trying to configure an inbound Xten softphone from outside. I have that user set as NAT in sip.conf (seems to help), but I still cannot establish a full session. I think the problem comes

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 1 going to another

Has anyone had any luck getting a ReplayTV DVR box to connect through an Asterisk box? Mine seems to dial just fine, but 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

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

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

First, the Asterisk settings: ----- sip.conf ----- [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) context=default ; Default context for incoming calls disallow=all ; First disallow all codecs; Set order allow=ulaw ; also known as g.711 PCMU; allow codecs in

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