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Are we interested, or ready for this? Andrew Seddon is offering to provide hardware to port Theora to the DSP Stamp. http://www.linuxdevices.com/news/NS4405077268.html John ---------- Forwarded Message ---------- Subject: RE: DSP stamp Date: Friday 29 October 2004 03:50 am From: "Andrew Seddon" <andrew.seddon@camsig.co.uk> To:...

...TFT hooked up for video out. So a good demo system would be to capture video from the camera chip and spew it out over the network post-compression (e.g a netcam!). The other way around could stream over the network and dump to the LCD. Drop me an e-mail if you are interested. Cheers, Andrew Seddon(andrew.seddon@camsig.co.uk) Cambridge Signal Processing Ltd. (www.camsig.co.uk) TEL: +44 1354 742563 FAX: +44 1354 740693 On Friday 29 October 2004 01:17 pm, you wrote: > If someone is willing to step up to do the port (are you volunteering > for this, John?), by all means this sounds like a...

...e some > rough performance figures for the theora codec, which I believe is based on > an earlier version of ON2's VP6 codec. > > Can I ask if you have a specific application in mind and if you are > interested in porting the theora codec? > > Best regards, > > Andrew Seddon(andrew.seddon@camsig.co.uk) > Cambridge Signal Processing, Ltd. (www.camsig.co.uk) > TEL: +44 1354 742563 > FAX: +44 1354 740693 > > > > -----Original Message----- > From: John Kintree [mailto:jkintree@swbell.net] > Sent: 28 October 2004 18:23 > To: sales@camsig.co.uk...

...t;<I am getting "Message: Originate failed" even the phone is ringing on the other end of the line.>> Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see. Bill Seddon ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Thomas Miller Sent: Mon 28/02/2005 17:33 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Manager "Message: Originate failed" beinggenerated when callee does not pick up I am using...

Thanks! Thanks! Thanks! I've got it work!!! :-) Message: 13 Date: Sun, 16 Jan 2005 12:17:21 -0000 From: "Bill Seddon" <bill.seddon@lyquidity.com> Subject: RE: [Asterisk-Users] failed to compile zaptel on redhat To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <ECOWS02MB8QDGMou3fY0007760a@smtp-out2.blueyonder.co.uk&...

Can anyone provide me with a step by step on how to set up Windows Messenger on a Windows XP Pro box as a SIP client with asterisk? I'm interested in doing various tests of my asterisk server from the Windows perspective of the world. In the alternative if someone could provide information on another Windows based fully functional easy to configure iax or SIP client that would suffice as

Folks, A few people have made me aware of some omissions in my files (not my fault, they weren't in the Script from the Wiki) which I shall be tackling this weekend. Whilst I'm making the files are there any other files you want? IVR's etc. If so make sure I have a script sent by email. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ

I'm trying to set a channel variable and make it available to another channel: I thought that if I SetVar(_SomeVariable=SomeValue) or SetVar(__SomeVariable=SomeValue) then SomeVariable would be available in the destination channel. However __SomeVariable, _SomeVariable and SomeVariable are all blank. The scenario: Agents logon to the queue using callbacklogin. From what I can gather

Hello all: Can anyone help me with finding the best locations for getting setup and other documentation for *. Thank you. Phil Menico www.xtend.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050110/6b26b235/attachment.htm

...starts. Such information could, of course, be stored in a database and perhaps that will be the long term objective. In the meantime I'm hoping that it is possible to use the built-in database and be able to run some kind of "autostart" context. Does such a facility exist? Bill Seddon

Hi all, Does anyone know how to get the channel ID on the other side of the call? For example: When SIP/50 calls SIP/21, and the call is answered by SIP/21 I get: SIP/21-6735 answered SIP/50-b456 ${CHANNEL} will show me SIP/50-b456. Is there a parameter or a workaround to get the SIP/21-6735 part? Thanks. Assaf Benharoosh -------------- next part -------------- An HTML attachment was

Hello, You can do either, you can send multiple Originate actions in a long line without waiting for a response back(although the responses do usually come back very fast) or you can open multiple connections using each one to Originate a new call. We use the multiple connection method in the astGUIclient suite because if you get a pause or lag in Manager output on a single connection(which does

Does anyone know if SIP will/is support on handheld PCs such as the iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems like a solution to provide a wireless based sip phone for any user would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can

I have started to receive a lot of positive response for the Asterisk@Home project. For those of you unfamiliar with this project the goal of Asterisk@Home is to make a full featured version of Asterisk very easy to install. We have created a 1 step .iso that installs RHEL (RedHat Enterprise Linux) and Asterisk. It includes a web GUI that allows easy editing of the Asterisk Config files.

...he words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C code available but if you already have the kit... Bill Seddon -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mark Phillips Sent: September 17, 2004 2:32 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] English vs American voice files Looks like I'...

Anybody know an easy way to adjust audio level of recordings made in Asterisk (using the 'record' application)? I've noticed that recordings using the "wav" format are about twice the level of those made using "WAV" or "wav49". Unfortunately, the "wav" recordings are uncompressed and about 10 times the size of the other formats. -brian

...disappeared with a cvs versions starting around 1.390 - 1.394 (chan_sip.c) according to my observations (more like a guess actually) couldn't exactly pinpoint the patch that stopped it. </QUOTE> My version of asterisk is from HEAD on 2004-09-19. Should I be concerned? Thanks Bill Seddon

Hello all! Is there any possibility to determine the IP address of a caller in my dialplan? I would like to have a predefined channel variable like ${CALLER_IP} but it seems it doesn't exist (http://www.voip-info.org/wiki-Asterisk+Variables) .. is this list complete? Are there any other possibility to store the SIP/IAX caller's IP address on every call? Thanks Niels

<< Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits.>> I have seen exactly this problem. Even IVR failed to work. Got an RMA from the supplier and they exchanged with no questions. Bill Seddon -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve Prior Sent: May 11, 2005 2:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIPURA SPA-2000 webserver dead after...

...;If you have done this any direction = you can offer will be appreciated.<o:p></o:p></span></font></p> <div> <p><font size=3D2 color=3Dnavy face=3DArial><span lang=3DEN-GB = style=3D'font-size:10.0pt; font-family:Arial;color:navy'>Bill Seddon</span></font><font = color=3Dnavy><span lang=3DEN-GB style=3D'color:navy'> </span><o:p></o:p></font></p> </div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span = style=3D'font-size:...