similar to: Where to download sample video file for Asterisk 1.8x playback?

Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the

Hi, I've a problem configuring my Asterisk. What I try to reach is to interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP) with 1 constraint I can't change : "every RTP flow needs to pass THROUGH Asterisk, and are NOT nated" What I observe : - a call made from a SIP Phone registred in Asterisk to Tandberg works (voice and video bidirectionnal) - a call

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in

Hello list, I'm trying to set up a video call from my Ekiga client to a Grandstream GXV3140 IP-phone. The call succeeds but there is no video. I have in sip.conf : videosupport=yes disallow=all allow=alaw allow=g726 allow=g729 allow=gsm allow=h261 allow=h263 allow=h263p allow=h264 The Grandstream peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 The Ekiga peer has codecs

I've tried to put a call file to /var/spool/asterisk/outgoing/ to make an outgoing video call, but not succeeded. I could hear the audio, but no video. The asterisk version is 1.4.10, with videosupport=yes The client is eyebeam 1.5.7, with h263 support. Here are some debug messages. It shows the client and asterisk negotiated the video capabilities without problem. However, the 'show

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

Hello All, I'am doing a video call between two Video Phones, and i see that Asterisk is stripping the fmtp parameters for the h263 video line in SDP. For example a line similar to the below is stripped, a=fmtp:xx CIF=4;QCIF=2;F=1;K=1 Asterisk is configured NOT to be present in the Media path (My version : Asterisk 1.4.19.1 ). I have the following enabled in my

supported by Asterisk for Video. I also find that video_caps branch has a fix for this problem, please can someone share more information about this and where i can find it ? I do not want those fmtp lines to be stripped. Suggestions to change the Asterisk config files, to achieve this are also welcome. Thank you. Best regards, Simith ------=_Part_17870_6007467.1218041254938 Content-Type:

I can create an audio call file and specify Application: Playback and Data: a path to the audio file, it calls the phone and plays the audio message just fine. I am trying to do the same with a video file. I specify Application: Playback and Data: the path to the video file (no ending of course), and I do specify also the Codecs: h264,h263 etc... Asterisk reports: *File /tmp/video does not

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 guys, I'm using Opus for speech in wide-band mode (sampling rate 16000) and 20ms frames with signal type set to SIGNAL_VOICE. I have a few questions here: 1. When I choose VBR mode, the codec seems to choose the bitrate on its own. However, that seems to be an issue on mobile devices. In some cases, when I configure the bitrate to say 20kbps, I see that the outgoing codec bitrate at

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

I've been using Linux Ubuntu for a few days, but I already love it :) However, when I'm running windows games under wine, I get less than 10fps, while under windows they work really fast. I have installed latest Nvidia drivers from ubuntu repositories and Direct X 9c under wine (yes, dxdiag works, and it passes all the tests). I have also seen on youtube, that many people don't have

Hi, I am currently using the following script to shape traffic on my linux box. I am getting ready to move the script to my linux firewall box so it shapes traffic for my home lan. I am looking to refine my rules a bit so to make bandwidth sharing a bit more efficient. Specifically I am looking to give out bound email more priority so that when a large email is sent, I want it to borrow all

How good is the video compression in vp3.2? For example, how does it compare (bit rate & quality) to mpeg1, mpeg2, Divx, real Video 8+, Windows Media, etc. etc. It being open source is nice and all, but if it can't be reasonably competitive then there isn't much point to it. I just haven't seen anything that compares vp32 to anything else. (Although to be honest, I haven't

The Opus RFC seems to recommend a frame size of 20ms for most applications. For wideband speech, the sweet spot range is recommended to be 16-20kbps. 20ms frames => 50 frames per second. For a VoIP application, the header overhead per frame (IP+UDP+RTP+SRTP) is 44bytes => 17.6kbps at 50 frames per second. So a 20ms frame size seems to cause a roughly 100% overhead of header data. Therefore,

Hi! It''s me again bothering you guys, what I want to do is to give full bandwidth to VPN traffic and limit the rest to 30KB/s (kilobytespersecond), ok? Here''s what I have: tcclasses ################################## eth0 1 1kbps 70kbps 1 eth0 2 1kbps 30kbps 2 default eth1 3 15kbps 10000kbps 1 eth1 4

Hello, I have been told on Freenode's #dovecot to ask here. I have Exim 4.92 complaining about "Non-CRLF-terminated header, under CHUNKING: message abandoned" with redirects made by Pigeonhole 0.5.5 (with Dovecot 2.3.5.1). Please find relevant part of the traffic below, and help me determine whether Exim is correct on this. ? 0x0000:? 4500 0954 0000 4000 8006 0000 7f00 0001? E..T..

Hi! I'm trying to run a game System Shock 2 http://appdb.winehq.org/appview.php?iVersionId=2270 I have a laptop with Celeron M 420 (1600Mhz). SS2 running something about 5-10fps.. opposing for example to DeusEx that runs _very_ smoothly.. I can say it is not less than 50fps or even more(i just don't know how to enable fps counter). Wine version is 0.9.35 Only messages i get from Wine