similar to: sipura fade to static

I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can't specify

Are there any known issues with call-waiting caller-id for SIP? Caller-ID on the first call works fine, but when the second call comes in, I hear the interrupt tone, but the caller-id doesn't display anything. I have tried this with the Cisco ATA and the SPA-2000. I have also tried two different phones to verify that it wasn't something specific to the phone. Thanks, Stephen

Can someone please explain what I am doing wrong here? I only want the extensions listed in long-users to be able to access the longdistance context. If I do this, I get a congestion tone no matter what I dial. If I add a [default] context and include => longdistance, then the local callers can call the long distance number fine, which is not what I want, but I still want long-users to be

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Has anyone succesfully integrated * with a cisco voice gateway ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/3b1ba7b3/attachment.htm

I have 2 questions regarding asterisk logs that I really hope someone can help me with. Jan 7 09:40:14 WARNING[1009517568]: File chan_iax.c, Line 3537 (registry_rerequest): Received unsolicited registry authenticate request from '209.242.15.34' I get this IAX message every minute or so. I have 2 asterisk servers that both register with each other. I can post the configuration again,

I''m pretty new to javascript, and scriptaculous, but I know what I want to do. I want to make it so that when a person clicks a link, the previous content of the div fades out, and the new content fades in. This is what I''ve got for my function: <script type="text/javascript"> //<![CDATA[ function loadmerch(im) { new

Hello, Has anyone had any good or bad experiences with the new Sipura 2.0 firmware? The 1.0.33 works pretty well but there are a few more features I'd like to have. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com

I have a list of categories, when I delete one of them, I want that item to fade and then get removed. So I do the following: 1. home_controller: def delete @category = Category.find_by_name(params[:name]) @element_id = @category.name Category.delete_all(["name = ?", @category.name]) end 2. delete.rjs: if @element_id page.visual_effect :fade, @element_id

I'm new to libtheora and video encoding in general, but I have worked hard to educate myself in the basics. I've working in image processing for many years, so I'm not starting entirely from scratch. I'm having an encoding problem and I'm looking for helpful suggestions. I'm using a very recent build of ffmpeg 0.6 to encode some image frames (+ audio) into

Hello. I was wondering if anybody could point me in the right direction of creating a cross-fade effect for Ajax.Updater. If you have an element that gets replaced by a new one with Ajax.Updater, how can we blend one into the other? Many thanks. Tench

Hi there, I''m now having an odd problem with Effect.Fade not working based on the innerHTML of a div. Take the following for example... <div id="blah"> Nothing here yet. </div> If I then update "blah", hide it and then fade it in, like so: <script type="text/javascript"> document.getElementById("blah").innerHTML =

Hi all! My first post here, so: be gentle ;-) I have been playing around with the scriptaculous library for a while now, and I must admit it is great. However, there is one thing that I''m notable to achieve: On Apple computers website for ilife (iphoto) http://www.apple.com/ilife/iphoto/ there is a verrrryyy nice fade effect going on when you click the navigation for ''new in

================================================= Mon-03-31-2008, 11:04pm U.S.EDT Hello, I''m trying to get a website entry page to work properly, while learning Scriptaculous effects and javascript at the same time. I wanted to use a couple of effects to add a little flash-style animation. I managed to get the initial Appear to work but with a problem: the image that''s supposed

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN

This patch adds mergeable receive buffer support to qemu-kvm, to allow enabling it when vhost_net supports it. It also adds a missing call to vhost_net_ack_features() to push acked features to vhost_net. The patch is relative to Michael Tsirkin's qemu-kvm git tree. +-DLS Signed-off-by: David L Stevens <dlstevens at us.ibm.com> diff -ruNp qemu-kvm.mst/hw/vhost_net.c

This patch adds mergeable receive buffer support to qemu-kvm, to allow enabling it when vhost_net supports it. It also adds a missing call to vhost_net_ack_features() to push acked features to vhost_net. The patch is relative to Michael Tsirkin's qemu-kvm git tree. +-DLS Signed-off-by: David L Stevens <dlstevens at us.ibm.com> diff -ruNp qemu-kvm.mst/hw/vhost_net.c

Hello, I currently have a problem : There is a domain administrators group which I filled as "admin users" in smb.conf. They can do what they want on files, that's fine. Problem is the files they create are owned by root. Let's explain why this is a problem for me : - People in this group can't access their files in an unix way (ftp, shell, ...) - If one of them get removed

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?