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  • Can I set up an "involuntary" conference call with Freeswitch?

    - by Atilla Filiz
    I am trying to set up a SIP/RTP public announcement infrastructure. Basically there are several slave user agents that are configured to answer automatically, and a master UA which should be able to call all of them and make announcements. A way to work around seems creating a conference and making all UAs to join via some RPC mechanism but I don't want to go that direction unless I have to. The slave UAs are linphone and I haven't decided on the master agent yet.

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  • Detecting 404 errors after a new site design

    - by James Crowley
    We recently re-designed Developer Fusion and as part of that we needed to ensure that any external links were not broken in the process. In order to monitor this, we used the awesome LogParser tool. All you need to do is open up a command prompt, navigate to the directory with your web site's log files in, and run a query like this: "c:\program files (x86)\log parser 2.2\logparser" "SELECT top 500 cs-uri-stem,count(*) FROM u_ex*.log WHERE sc-status=404 GROUP BY cs-uri-stem order by count(*) desc" -rtp:-1 topMissingUrls.txt And you've got a text file with the top 500 requested URLs that are returning 404. Simple!

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  • FREE Windows Azure Boot camp &ndash; Raleigh Wednesday June 23, 2010

    - by Jim Duffy
    Just want to be sure you don’t miss out on an opportunity to take advantage of some free Windows Azure training. Microsoft Developer Evangelist Brian Hitney and I will be presenting a one-day Windows Azure boot camp on June 23rd in Raleigh, NC at the Microsoft RTP offices. For more information on content, what to bring, directions, etc. just click here to go to the information and registration page for the Raleigh event. To find other dates and locations for the Windows Azure boot camps  head over to the Windows Azure Boot Camp page. Brian and I hope to see you there! Have a day. :-|

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  • Windows Azure Boot camp &ndash; Raleigh Wednesday June 23, 2010 * FREE*

    - by Jim Duffy
    Yes I know this is my second blog post about the free one-day Windows Azure boot camp on June 23rd in Raleigh, NC. What can I say I don’t want anyone to miss out on an opportunity to take advantage of some free Windows Azure training. Microsoft Developer Evangelist Brian Hitney and I will be presenting a one-day Windows Azure boot camp on June 23rd in Raleigh, NC at the Microsoft RTP offices. For more information on content, what to bring, directions, etc. just click here to go to the information and registration page for the Raleigh event. To find other dates and locations for the Windows Azure boot camps  head over to the Windows Azure Boot Camp page. Brian and I hope to see you there! Have a day. :-|

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  • How can I force the (re)discovery of PulseAudio network sound devices?

    - by Christian
    I'm using the PulseAudio feature of network sound devices (not Multicast/RTP) to play sound from my netbook on the audio equipment connected to the HTPC when at home. This creates a virtual sound device that I can then use instead of the physical built-in one. Most of the time this works just fine. Sometimes however, the virtual sound device just doesn't appear. Disconnecting from and reconnecting to the network helps sometimes but not always and it's annoying and potentially bad for existing TCP connections. So my question basically is: Is there some way to tell PulseAudio "Hey, just look again if you really can't find a network sound device."? Edit: Unloading and reloading the module-zeroconf-discover with pacmd does not help either and it doesn't appear to be an avahi problem per se since avahi-browse -t --all | grep PulseAudio shows lots of right-looking stuff, even when the devices aren't listed in pavucontrol or pacmd list-sinks. Edit 2: I'm using Ubuntu 12.04 on both boxes for all the difference it might make.

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  • RTSP client in android

    - by Vinay
    I am writing a RTSP client in Android. I am able to receive the Responses for all the requests i.e., DESCRIBE it sends back the 200 OK SETUP with transport: RTP/AVP:unicast:client_port=4568:4569 got the 200 OK Message back Sent PLAY, and got the OK Message After that how to get the audio and video frames? I have searched on blogs, but all say to listen at client_port but I am not receiving any packets. Please let me know am I doing correctly.

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  • JMF. Create new custom streamdatasource

    - by Afro Genius
    Hi there. I am looking to create a means of building a DataSource object (and hence a Processor) that gets data from a stream instead of a file, RTP, and so on. I am writing a module for a much larger application that is meant to transparently transcode audio data. Going through the JMF docs only specify how to create a source from file however I need to be able to create a source from a stream within my application. Any idea where I can start looking?

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  • Running a small IPTV station

    - by nixterrimus
    I'm looking to run an iptv station for my dorm. I know I can serve multicast so that's not a problem. The station will serve out podcasts and other cc licensed content. The target endpoint is xbmc- a media center. So far I know that I need to serve an rtp stream over udp that's streaming an mpeg-4 avc main or high profile with aac ( or ac3 ?) audio. I've had some luck using vlc with vlm to stream but it seems limited. What are my other options?  Everything has to run on Linux- hopefully open source. How can I use playlists and not live streams? What are my software options?

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  • ffserver - streaming problem transcodation for input

    - by zozo
    Good day to all. I have a little problem. I'm trying to stream something from a cam to a server and then forward to... somewhere (it will be a site or something). On the computer that I have the cam connected to I use vlc to stream it to the server and there I try to get the stream as an input for a ffserver. The problem is that ffserver doesn't detect the input (regardless of the protocol I use (udp, rtp, etc.)). I suspect a transcoding problem or something like that but I can't find any documentation about that so... Does any1 know what transcodation I should use? Thank you for help and have a great day.

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  • Understanding Asterisk Server features

    - by Arham Ali Qureshi
    I need to ask few question about Asterisk 1) Does ACL mean by Access Control list here ?If yes than how could i use it? >ip show user 6001 * Name : 6001 Secret : <Set> MD5Secret : <Not set> Context : DLPN_Admin Language : AMA flags : Unknown Transfer mode: open MaxCallBR : 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 2147483647 Callgroup : 1 Pickupgroup : 1 Callerid : "test" <6001> ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No 2) What is mean by "Require Call Token" in Asterisk Digium GIU on Create new User Panel 3) Is There any command from where i can get users VOICE MAIL password ? 4) What AMI or CLI command set call recording on or off for user ? and if i want that file to be stored on client computer not on server memory what could i do ?

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  • How to generate a 8 bit per sample wav audio file in VLC

    - by Ahmed safan
    I'm using the following vlc command line to extract first 5 minutes of audio from video file "-I dummy -vvv --no-sout-video --sout-audio --no-sout-rtp-sap --no-sout-standard-sap --ttl=1 --sout-transcode-threads=5 --sout-transcode-high-priority --sout-keep --sout #transcode{acodec=s16l,channels=1,samplerate=8000,ab=64}:std{mux=wav,access=file,dst="c:\dest.wav"} "c:\originalvideo.mpg" --start-time=0 --stop-time=300 vlc://quit"; if ab=64 =64 k bits per second and samples per second=8 k samples then bits per sample=64/8=8 bits per sample but the problem is that the output file always has samples of 16 bits per sample. I know that sample can contain bits from 8 , 16, 24 to 32 bits per sample. i want to get 8 bits per sample file how can this be done ?

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  • Using VLC to Unicast High Definition Webcam over local gigabit LAN with low/zero delay

    - by Robin Day
    We're setting up a webcam "window" between two offices in the same buildilng. The two PC's are connected to the same gigabit switch. We're using VLC to stream the webcam over HTTP using the following commands. vlc dshow:// :dshow-caching="0" :dshow-size="640x480" :sout=#transcode{vcodec=h264,vb=0,scale=0}:http{mux=ffmpeg{mux=flv},dst=:8080/} :no-sout-rtp-sap :no-sout-standard-sap :ttl=1 :sout-keep vlc http://192.168.0.1:8080 :http-caching="0" Even with the caching set to zero, the delay in the image is a good 2-3 seconds. The CPU usage of each pc is also maxed. I'm guessing it's the transcoding that's causing much of the delay. Can anyone give me some changes to these command lines that will reduce the transcoding power, or send the webcam over a different protocol, or anything that will reduce the delay of the cameras? Bandwidth is not an issue at all as the pc's can be connected to a dedicated switch/vlan if required.

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  • Lots of artifacts while streaming HD content with VLC 0.9.9 on CentOS

    - by Zsub
    I'm trying to stream (multicast) a x264 encoded file using VLC. This in itself succeeds, but the stream has a huge lot of artifacts. This seems to suggest that the data cannot be transported fast enough. If I check network usage, though, it's only using about 15 mbit. I have a similar SD stream which functions perfectly. I think I could improve stream performance by not streaming the raw data, but I cannot seem to get this working. It seems that on keyframes all artifacts are removed for a short while (less than a second). This is the command I use: vlc -vv hdtest.mkv --sout '#duplicate{dst=rtp{dst=ff02::1%eth1,mux=ts,port=5678,sap,group="Testgroup",name="TeststreamHD"}}' --loop Which is all one long line.

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  • Getting RINGING response on SIP UAC without sending it from the other UAC

    - by TacB0sS
    Hi, I hope this would be my last question about this SIP subject, I have managed to overcome the last issue I had by asking a friend to help me from a remote computer, I'm able to connect between the computers, but here is the thing, according to all the examples I saw, the Callee should invoke the Ringing response, but in my application case I didn't implement it yet, but I still receive on the Caller UAC a Ringing response, this is the SIP messages that are on the caller end: Outgoing Request 5: INVITE sip:[email protected] SIP/2.0 Contact: "Client 310" <sip:[email protected]> From: "Client 310" <sip:[email protected]> Max-Forwards: 32 CSeq: 2 INVITE Call-ID: [email protected] Allow: INVITE,CANCEL,ACK,BYE,OPTIONS Content-Type: application/sdp Proxy-Authorization: Digest username="310",nonce="012afffb",realm="asterisk",uri="sip:[email protected]",algorithm=MD5,response="d19ca5b98450b4be7bd4045edb8a3a2f" Via: SIP/2.0/UDP hostName.hn:5060 To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Content-Length: 257 v=0 o=310 7108915969559970847 7108915969559970847 IN IP4 xxx.xxx.x.xxx s=- i=Nu-Art Software - TacB0sS VoIP information c=IN IP4 xxx.xxx.x.xxx m=audio 3312 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Incoming Response 6: SIP/2.0 100 Trying Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 Incoming Response 7: SIP/2.0 180 Ringing Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 Call to: [email protected] is Ringing Incoming Response 8: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 264 v=0 o=root 27669 27669 IN IP4 yy.yy.yy.yy s=session c=IN IP4 yy.yy.yy.yy t=0 0 m=audio 10914 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Incoming Response 9: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Content-Length: 0 I do not respond to the invite, that is why all this is happening, but why am I getting a ringing if I'm not the one sending it. Thanks, Adam.

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  • Windows 8 Developer Camp - Raleigh September 25th

    - by Jim Duffy
    Time is ticking away and the time to act is now! How's that for some motivation? :-)  Microsoft Developer Evangelist Brian Hitney and I want to help you get your app in the store in time for the October 26 Windows 8 launch. Come join us on Tuesday, September 25, at 9:00 AM in the Microsoft RTP offices to learn how simple it can be to construct a world class Windows 8 application. Don't think you can be ready to join the Windows 8 launch? Come anyway. You might be surprised. Don't have any idea what kind of app to build? Come anyway. They're are plenty of places to look for inspiration. Either way you can learn what it takes to create or tune an app for Windows 8 and publish it in the Windows App Store. Learn more about this free event on the registration page. Registration is now open and space is limited. Have a day.

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  • DirectShow - passing parameters to custom source push filter

    - by mkurek
    Hello, I'm working on a solution that will be used to receive video stream from remote hosts and to put various texts on the top of it. Currently it consists of custom DirectShow push filter (C++) which receives data from remote hosts using RTP protocol and tiny C# application that sets up the DirectShow graph and is used as a container for the video. I'm using DirectShowLib interop library. However, I'm not sure how to pass parameters from this C# app to my custom filter. What are possible ways to do it?

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  • streaming to correct network interface

    - by robin hood
    I have IP cam that supports RTSP streaming. It's connected to router with 2 network cards with IP1 and IP2 addresses. I make 2 connections to IP cam by IP1 and IP2 addresses from the same IP and I need to receive corresponding streams thru correct network card, but both streams (RTP over UDP) go thru IP1. How this can be resolved? I don't know if RTSP server binds UDP sockets to corresponding IP and I don't know what IP stack is in IP cam (weak end system or strong end system). I haven't found anything interesting in router configuration. As I understand, routing table cannot help me cos I'm connected from the same IP, is it right? Also Sorry for incomplete info but it's all I have at the moment. Thanks for your time.

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  • how to configure IP cam to stream using right network card?

    - by robin hood
    I have IP cam that supports RTSP streaming. It's connected to router with 2 network cards with IP1 and IP2 addresses. I make 2 connections to IP cam by IP1 and IP2 addresses from the same IP and I need to receive corresponding streams thru correct network card, but both streams (RTP over UDP) go thru IP1. How this can be resolved? I don't know if RTSP server binds UDP sockets to corresponding IP and I don't know what IP stack is in IP cam (weak end system or strong end system). I haven't found anything interesting in router configuration. As I understand, routing table cannot help me cos I'm connected from the same IP, is it right? Also Sorry for incomplete info but it's all I have at the moment. Thanks for your time.

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  • gstreamer pulseaudio echo cancellation

    - by user3618055
    I'm implementing a voip application using gstreamer, i use the example of the rtp in the plugin-good! i want to implement echo cancellation, i couldn't use the speex echo canceller with gstreamer because the input and the output are not in the same process. So, i want to use pulse audio to make echo cancellation? can any one help me how to deal with? the sender voice is pipeline = gst_pipeline_new (NULL); g_assert (pipeline); /* the audio capture and format conversion */ audiosrc = gst_element_factory_make (pulsesrc, "audiosrc"); g_assert (audiosrc); audioconv = gst_element_factory_make ("audioconvert", "audioconv"); g_assert (audioconv); audiores = gst_element_factory_make ("audioresample", "audiores"); g_assert (audiores); /* the encoding and payloading */ audioenc = gst_element_factory_make (AUDIO_ENC, "audioenc"); g_assert (audioenc); audiopay = gst_element_factory_make (AUDIO_PAY, "audiopay"); g_assert (audiopay); /* add capture and payloading to the pipeline and link */ gst_bin_add_many (GST_BIN (pipeline), audiosrc, audioconv, audiores, audioenc, audiopay, NULL); if (!gst_element_link_many (audiosrc, audioconv, audiores, audioenc, audiopay, NULL)) { g_error ("Failed to link audiosrc, audioconv, audioresample, " "audio encoder and audio payloader"); } and the receiver is : gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL); /* the depayloading and decoding */ audiodepay = gst_element_factory_make (AUDIO_DEPAY, "audiodepay"); g_assert (audiodepay); audiodec = gst_element_factory_make (AUDIO_DEC, "audiodec"); g_assert (audiodec); /* the audio playback and format conversion */ audioconv = gst_element_factory_make ("audioconvert", "audioconv"); g_assert (audioconv); audiores = gst_element_factory_make ("audioresample", "audiores"); g_assert (audiores); audiosink = gst_element_factory_make (pulsesink, "audiosink"); g_assert (audiosink); /* add depayloading and playback to the pipeline and link */ gst_bin_add_many (GST_BIN (pipeline), audiodepay, audiodec, audioconv, audiores, audiosink, NULL); res = gst_element_link_many (audiodepay, audiodec, audioconv, audiores, audiosink, NULL); g_assert (res == TRUE); i tried to change gstreamer proprietes to pulseaudio server in input and output and i used "pactl load-module module-echo-cancel aec_method=adrian" but i still listen to echo!! any one could help please thanks!!

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  • Live Screencast under Linux

    - by OmnipotentEntity
    I was having some difficulty with running a Live Screencast under Linux. I've found jtvlc and tried using that, but whenever I use it the stream comes out either blank or lagged with extremely high latency. I have a fast internet connection and a fast computer, but am I perhaps taxing it too much? Any ideas on what I could possibly be doing wrong? # 1. Get an account on http://www.justin.tv/ # 2. Copy streaming key from: http://www.justin.tv/broadcast/adv_other # 2. Install VLC: http://www.videolan.org/vlc/ # 3. Get Win/Mac/Lin Stream Client: \ # http://apiwiki.justin.tv/mediawiki/index.php/Linux_Broadcasting_API # 4. Adjust the vlc parameters to your liking and run VLC like this #!/bin/bash cvlc screen:// --input-slave=pulse:// \ --screen-width 1920 \ --screen-height 1080 \ --screen-fps 5 \ -v input_stream \ --sout='#duplicate{ dst="transcode{ scale=1, venc=x264{ keyint=60 }, vcodec=h264, vb=600, acodec=mp4a, ab=32, channels=2, samplerate=22050 } :rtp{dst=127.0.0.1,port=1234,sdp=file:///tmp/vlc.sdp} "}' \ --sout-transcode-threads=4 & sleep 2 # 5. Run JTVLC to stream like this: ./jtvlc/jtvlc omnipotententity censored /tmp/vlc.sdp # Notes: #- If you want to see what you're about to stream add 'dst=display, ' # before 'dst="transcode[' # More about the VLC parameters: http://wiki.videolan.org/Documentation:Modules/screen

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  • PSTN Trunk TDM400P Install on Asterisk / Trixbox

    - by Jona
    Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I've run setup-pstn which produces the following output trixbox1.localdomain ~]# setup-pstn -------------------------------------------------------------- Detecting PSTN cards and USB PSTN Devices -------------------------------------------------------------- Hardware present! STOPPING ASTERISK Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: [ OK ] wcte12xp: [ OK ] wct1xxp: [ OK ] wcte11xp: [ OK ] wctdm24xxp: [ OK ] opvxa1200: [ OK ] wcfxo: [ OK ] wctdm: [ OK ] wcb4xxp: [ OK ] wctc4xxp: [ OK ] xpp_usb: [ OK ] Running dahdi_cfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started Chan Extension Context Language MOH Interpret Blocked State pseudo default en default In Service 1 from-pstn en default In Service dahdi_scan returns: dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 00 Slot 10 basechan=1 totchans=4 irq=209 type=analog port=1,FXO port=2,none port=3,none port=4,none And asterisk can see the channel: > trixbox1*CLI> dahdi show channel 1 > Channel: 1LI> File Descriptor: 14 > Span: 11*CLI> Extension: I> Dialing: > noI> Context: from-pstn Caller ID: I> > Calling TON: 0 Caller ID name: > Mailbox: none Destroy: 0LI> InAlarm: > 1LI> Signalling Type: FXS Kewlstart > Radio: 0*CLI> Owner: <None> Real: > <None>> Callwait: <None> Threeway: > <None> Confno: -1LI> Propagated > Conference: -1 Real in conference: 0 > DSP: no1*CLI> Busy Detection: no TDD: > no1*CLI> Relax DTMF: no > Dialing/CallwaitCAS: 0/0 Default law: > ulaw Fax Handled: no Pulse phone: no > DND: no1*CLI> Echo Cancellation: > trixbox1128 taps trixbox1(unless TDM > bridged) currently OFF Actual > Confinfo: Num/0, Mode/0x0000 Actual > Confmute: No > Hookstate (FXS only): Onhook A cat of /etc/asterisk/dahdi.conf shows: [trixbox1.localdomain ~]# cat /etc/asterisk/dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Tue May 25 17:45:13 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) ;;; line="1 WCTDM/4/0 FXSKS (SWEC: MG2)" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 1 callerid= group= context=default I have configured a "ZAP Trunk (DAHDI compatibility Mode)" with the ZAP identifier 1 and an outbound route, but when ever I try to make an external call via it I get the "All Circuits are busy now, please try your call again later message". I have one outbound route which uses the dial pattern 9|. and the Trunk Zap/1 and one Zap Trunk which uses Zap Identifier (trunk name): 1 and has no Dial Rules. The FXO module is directly connected to our phone line from BT via a BT-RJ11 cable. When running tail -f /var/log/asterisk/full and placing a call I get the following output: [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP TOS bits 184 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP RTP CoS mark 5 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP TOS bits 136 [May 26 11:10:52] VERBOSE[2723] logger.c: == Using SIP VRTP CoS mark 6 [May 26 11:10:52] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:1] Macro("SIP/801-b7ce8c28", "user-callerid,SKIPTTL,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:1] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/801-b7ce8c28", "1?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:4] Set("SIP/801-b7ce8c28", "AMPUSER=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:5] Set("SIP/801-b7ce8c28", "AMPUSERCIDNAME=Jona") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/801-b7ce8c28", "0?report") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:7] Set("SIP/801-b7ce8c28", "AMPUSERCID=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:8] Set("SIP/801-b7ce8c28", "CALLERID(all)="Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:9] Set("SIP/801-b7ce8c28", "REALCALLERIDNUM=801") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:10] ExecIf("SIP/801-b7ce8c28", "0?Set(CHANNEL(language)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:11] GotoIf("SIP/801-b7ce8c28", "1?continue") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-user-callerid,s,20) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-user-callerid:20] NoOp("SIP/801-b7ce8c28", "Using CallerID "Jona" <801>") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:2] Set("SIP/801-b7ce8c28", "_NODEST=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:3] Macro("SIP/801-b7ce8c28", "record-enable,801,OUT,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/801-b7ce8c28", "1?check") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-record-enable,s,4) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/801-b7ce8c28", "recordingcheck,20100526-111052,1274868652.1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [May 26 11:10:52] VERBOSE[2858] logger.c: recordingcheck,20100526-111052,1274868652.1: Outbound recording not enabled [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28>AGI Script recordingcheck completed, returning 0 [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-record-enable:5] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:4] Macro("SIP/801-b7ce8c28", "dialout-trunk,1,01483890915,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/801-b7ce8c28", "DIAL_TRUNK=1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/801-b7ce8c28", "0?sub-pincheck,s,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/801-b7ce8c28", "0?disabletrunk,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/801-b7ce8c28", "DIAL_NUMBER=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=tr") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/801-b7ce8c28", "OUTBOUND_GROUP=OUT_1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/801-b7ce8c28", "1?nomax") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s,9) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/801-b7ce8c28", "0?skipoutcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/801-b7ce8c28", "DIAL_TRUNK_OPTIONS=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/801-b7ce8c28", "outbound-callerid,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/801-b7ce8c28", "0?Set(REALCALLERIDNUM=801)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/801-b7ce8c28", "1?normcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,6) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/801-b7ce8c28", "USEROUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/801-b7ce8c28", "EMERGENCYCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/801-b7ce8c28", "TRUNKOUTCID=") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/801-b7ce8c28", "1?trunkcid") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-outbound-callerid,s,12) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERID(all)=)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/801-b7ce8c28", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/801-b7ce8c28", "0?AGI(fixlocalprefix)") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/801-b7ce8c28", "OUTNUM=01483890915") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/801-b7ce8c28", "custom=DAHDI/1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/801-b7ce8c28", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/801-b7ce8c28", "dialout-trunk-predial-hook,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/801-b7ce8c28", "0?bypass,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/801-b7ce8c28", "0?customtrunk") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/801-b7ce8c28", "DAHDI/1/01483890915,300,") in new stack [May 26 11:10:52] WARNING[2858] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [May 26 11:10:52] VERBOSE[2858] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-dialout-trunk:20] Goto("SIP/801-b7ce8c28", "s-CHANUNAVAIL,1") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/801-b7ce8c28", "1?noreport") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/801-b7ce8c28", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [901483890915@from-internal:5] Macro("SIP/801-b7ce8c28", "outisbusy,") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:1] Playback("SIP/801-b7ce8c28", "all-circuits-busy-now,noanswer") in new stack [May 26 11:10:52] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'all-circuits-busy-now.ulaw' (language 'en') [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-outisbusy:2] Playback("SIP/801-b7ce8c28", "pls-try-call-later,noanswer") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- <SIP/801-b7ce8c28> Playing 'pls-try-call-later.ulaw' (language 'en') [May 26 11:10:54] WARNING[2661] pbx.c: FONALITY: This thread has already held the conlock, skip locking [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/801-b7ce8c28' in macro 'outisbusy' [May 26 11:10:54] VERBOSE[2858] logger.c: == Spawn extension (from-internal, 901483890915, 5) exited non-zero on 'SIP/801-b7ce8c28' [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [h@from-internal:1] Macro("SIP/801-b7ce8c28", "hangupcall") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/801-b7ce8c28", "vw") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("SIP/801-b7ce8c28", "1?skiprg") in new stack [May 26 11:10:54] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,6) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("SIP/801-b7ce8c28", "1?skipblkvm") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,9) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("SIP/801-b7ce8c28", "1?theend") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: -- Goto (macro-hangupcall,s,11) [May 26 11:10:55] VERBOSE[2858] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("SIP/801-b7ce8c28", "") in new stack [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/801-b7ce8c28' in macro 'hangupcall' [May 26 11:10:55] VERBOSE[2858] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/801-b7ce8c28' I'm guessing I've missed a configuration step somewhere but no idea where, any help greatly appreciated.

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  • Hey Guy , I want ot streaming video by using VideView class . Can anyone tell me what format is it s

    - by eddyxd
    Hi , I am the newbie of android, but i hava seen the tutorial and implement some simple applications. The question i met is that I am tring to stream some video from my server to android, but the android VideoView class just plays the audition sololy without "image"@@!~ Here is my setting and android code : 1. android core code: mVideoView01.setVideoURI(Uri.parse("rtsp://192.168.16.1:8080/test.sdp")); mVideoView01.start(); 2. my streaming server is VLC and the command is: vlc -vvv d:\nobody.mp4 --sout=#transcode{vcodec=h264,width=320,hegiht=240}:rtp{dst=192.168.16.1,port=4444,sdp=rtsp://192.168.16.1:8080/test.sdp} ps: My ip is got from DHCP but I have checked it really can be connected(Android could play audition after all) ps2: I haved trid to stream some video from "http://www.americafree.tv/" and the playing is good!!@@ So I guess that the problem maybe is caused by streaming Video format, but I have almost tried every figument option form VLC, and it still don't workQQ. So Have anyone done the same test as me can give me some advice?? Thanks a lot!!!!! by eddy

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  • moving audio over a local network using GStreamer

    - by James Turner
    I need to move realtime audio between two Linux machines, which are both running custom software (of mine) which builds on top of Gstreamer. (The software already has other communication between the machines, over a separate TCP-based protocol - I mention this in case having reliable out-of-band data makes a difference to the solution). The audio input will be a microphone / line-in on the sending machine, and normal audio output as the sink on the destination; alsasrc and alsasink are the most likely, though for testing I have been using the audiotestsrc instead of a real microphone. GStreamer offers a multitude of ways to move data round over networks - RTP, RTSP, GDP payloading, UDP and TCP servers, clients and sockets, and so on. There's also many examples on the web of streaming both audio and video - but none of them seem to work for me, in practice; either the destination pipeline fails to negotiate caps, or I hear a single packet and then the pipeline stalls, or the destination pipeline bails out immediately with no data available. In all cases, I'm testing on the command-line just gst-launch. No compression of the audio data is required - raw audio, or trivial WAV, uLaw or aLaw encoding is fine; what's more important is low-ish latency.

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  • libpcap read packet size

    - by spicyramen
    I started to write an application which will read RTP/H.264 video packets from an existing .pcap file, I need to read the packet size. I tried to use packet-len or header-len, but it never displays the right number of bytes for packets (I'm using wireshark to verify packet size - under Length column). How to do it? This is part of my code: while (packet = pcap_next(handle,&header)) { u_char *pkt_ptr = (u_char *)packet; struct ip *ip_hdr = (struct ip *)pkt_ptr; //point to an IP header structure struct pcap_pkthdr *pkt_hdr =(struct pcap_pkthdr *)packet; unsigned int packet_length = pkt_hdr->len; unsigned int ip_length = ntohs(ip_hdr->ip_len); printf("Packet # %i IP Header length: %d bytes, Packet length: %d bytes\n",pkt_counter,ip_length,packet_length); Packet # 0 IP Header length: 180 bytes, Packet length: 104857664 bytes Packet # 1 IP Header length: 52 bytes, Packet length: 104857600 bytes Packet # 2 IP Header length: 100 bytes, Packet length: 104857600 bytes Packet # 3 IP Header length: 100 bytes, Packet length: 104857664 bytes Packet # 4 IP Header length: 52 bytes, Packet length: 104857600 bytes Packet # 5 IP Header length: 100 bytes, Packet length: 104857600 bytes Another option I tried is to use: pkt_ptr- I get: read_pcapfile.c:67:43: error: request for member ‘len’ in something not a structure or union

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  • JMF microphone volume controller

    - by TacB0sS
    How to obtain the Microphone volume controller in JMF? this is what I have: I tried this implementation concept of yours, but I keep getting a null from the first volume processor when I try to get the stream, here is how I do it: // the device is the media device specifically audio Processor processorForVolume = Manager.createProcessor(device.getLocator()); // wait until configured ProcessorStates newState = new ProcessorStateListener(Processor.Configured).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // setting the content descriptor to null - read in another thread this allows to get the gain control processorForVolume.setContentDescriptor(null); // set the track control format to one supported by the device and the track control. // I didn't match it to an RTP allowed format, but I don't think this has anything to do with it... TrackControl[] trackControls = processorForVolume.getTrackControls(); if (trackControls.length == 0) throw new MC_Exception("No track controls where found for this device:", new Object[]{device}); for (TrackControl control : trackControls) trackManipulator.manipulateTrackControls(control); // wait until the processor is realized newState = new ProcessorStateListener(Controller.Realized).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // receives the gain control micVolumeController = processorForVolume.getGainControl(); // cannot get the output stream to process further... any suggestions? processor = Manager.createProcessor(processorForVolume.getDataOutput()); new ProcessorStateListener(Processor.Configured).waitForProcessorState(processor); processor.setContentDescriptor(DeviceCapturingManager.RAW_RTP); new ProcessorStateListener(Controller.Realized).waitForProcessorState(processor); this is the output It generates: volumeProcessorState: Configured format set to track control - com.sun.media.ProcessEngine$ProcTControl@1627c16: LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed volumeProcessorState: Realized and the data output from the processor is Null. I should make clear that when the content descriptor != null I do get an output stream but not the volume controller, and the when it is null I get the controller, but no stream. I try to connect to an audio microphone device Adam.

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