similar to: * Video changes

was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286

I''m trying to build a selection list which I have done in various ways but this one is new to me. I have a ''facilities'' table which has all the outpatient facilities but I need to add ''Float'' and ''Main Office'' which I don''t want to add to the ''facilities'' table itself. so I figure I can add these to an

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

I am trying to set up a PDC controller on a samba server, but continue to get the following error: The user could not be added because the following error occured: The trust relationship between the workstation and the primary domain failed. An extract from the log shows only the following: [2003/08/01 22:58:42, 0] smbd/service.c:make_connection(381) make_connection: bkruger logged in as

Using rsync-2.4.6: Is a times difference supposed to cause a write? Also -t vs -I makes no difference. Below shows the problem, I think: [dmahurin@pc16 /tmp]$ mkdir x y [dmahurin@pc16 /tmp]$ cp /bin/ls x [dmahurin@pc16 /tmp]$ ls -l x/ls -rwxr-xr-x 1 dmahurin users 43024 Nov 13 12:46 x/ls [dmahurin@pc16 /tmp]$ rsync -vrtW x/ y building file list ... done ./ ls ./ wrote 43112 bytes

Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the "v" flag on my extension for the meetme app? Thanks, Tim

I'd like to cobble together a videophone from an analog phone, connected to an Asterisk FXS channel, and a co-located video camera, connected to a video grabber card on the Asterisk server (so I have a Linux video device providing the video stream). When a call is made from the phone, I'd like to somehow add the video and produce a SIP video call. I don't want to use any sort of

Hello, can anyone comment on how one could use SIPphone's $89 All-in-One adapter with Asterisk? Sounds to me like it should work as both a FXO and FXS. It would be a cheap way of getting started with Asterisk and PSTN. Any comments on the SIPphone FX200? Any comments on SIPphone in general? Thank you for your help

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

I have a lot of experience, all of it pretty good, with various Sipura products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into Asterisk as clients. Good sound quality, great reliability. I've tried two of the units named in the subject line, and frankly I'm frustrated. Calls usually start out OK, but within a brief period the sound goes totally to

Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I

I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed

Hi, My goal is to set up the server so that one directory acts like a windows share that (1) does not require any log in information to gain access (2) Can be viewed from a windows box and selected using map network drive. At the the same time, I also want to set up private space on the disk that does require an authorized user, username and password for access. My set up (see smb.conf

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3)

Hello, just a short hardware question. Does CentOS 5.3 supports a Leadtek LR2960 (model S26361-D1910-V128, agp, 128mb) graphic card with a nVidia GeForce FX5200 chip and dual dvi? Thank you very much! regards Olaf

I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based

Hi, One of my clients, wants to use * box to run weekly meetings between remote locations over the internet. What would be the best configuration for this? We are talking about two conference rooms. I am referring to the actual hardware/software and bandwidth requirements for this to work well. I have run two software video phones and I had marginal results with it when displayed on large LCDs,

When I start * I get 6 mpg123 processes start as well. Is this normal? Often after a couple of days these mpg123 processes start to consume cpu and I have to kill them off. I do not have a sound card in the server and I have noload => chan_oss.so Simon

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic

Hi, I've gotten hold of a few SIP videophones and would love to be able to transfer the video-feed from RTP through IAX2 which is used between two asterisk boxes: Videophone (SIP) -> * (IAX2) -> * (SIP) -> Videophone I did get nice video between the two phones when they were registered to the same * box, but over IAX2 it won't work. Can it be done ? (ChangeLog says there is