similar to: rtsp help

Hello, I am trying to stream an IP Camera output (h264) into a conference. The IP Camera supports RTSP. Searching around the web, I believe the RTSP support (was) available through app_rtsp (external to Asterisk distribution). This, I believe, has problems and has issues compiling in Asterisk 11 (I tried compiling it in Asterisk 11 and it failed). I may not be able to use DiaStar or i6net's

Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrakora at messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032

I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. "dialplan show" gives me that the context is present: [ Context 'smvoice-mediaport' created by

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

Hi all, I have setup my Cisco 79XX phone. Did the tftp, put the config files in the right location with the right names. Booted my phone, it does the tftp things, the screen shows my extensions everything seems fine. However, when I come offhook and try to dial 11 which is just a playback of demo-congrats in the dialplan the phone says Calling Out (INV) below is my sip.conf file - I presume it

I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in

Some progress to report: I had to run asterisk as the user logged in - actually not even that. I could not "su user -c " to that user - I had to run it as that user. Then I did a test and got audio! Great... But when I do a second test. Asterisk HANGS on ChanIsAvail() Then I thought lets SKIP that - and just let it do the Dial() - I stopped everything - got it running again. - and

Greetings: It would be nice if Icecast supported RTSP; however I would appreciate any suggestions for a small RTSP/RTP solution to encode 8kHz mono audio in GSM or ADPCM and service multiple unicast client connections. The ideal would be a black-box hardware solution with an audio input and ethernet interface similar to broadcast studio IP audio links or the network audio capabilities of certain

Hi Where do I find instruction to configure icecast as a reflect server for rtsp? Scenario: Client1 sends audio to IceCast via rtsp Client2 receives audio from IceCast via rtsp. I need to just setup this behaviour.. -Atul -- ------- To make world a better place to live in ------- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I''m running performance tests on a Linux router using IPTables to nat traffic over the network. I have a MS Media streamer, and two windows clients behind the router which download video from the streamer. I can then measure performance. MS Media player rolls amongst protocols until it finds one it can use as follows RTSP UDP RTSP TCP MMS UDP MMS TCP HTTP Unfortunately it

I am running 1.4.22. I am doing a simple call into the dialplan and am getting a strange error: [Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user "404" <sip:404 at 192.168.1.8>;tag=547521CB-DB0D6130 This is the only line that prints on the console... Typically I get a few lines like: -- Executing [33 at smvoice-sip:1]

I am using asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically "core show channels concise" "sometimes" I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF ]' I'm not

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

Thomas B. Ruecker wrote: > Michael Grigoni wrote: > >>Greetings: >> >>It would be nice if Icecast supported RTSP; > > It probably never will > >>however I would >>appreciate any suggestions for a small RTSP/RTP solution to >>encode 8kHz mono audio in GSM or ADPCM and service multiple >>unicast client connections. > > why not use

Hi, I have been trying to use wine to get at a streaming quicktime server on Ubuntu 8.04, but as of yet no luck: The server streams mov files over RTSP First I tried using just VLC to play it, but there are some things it just can't handle (in windows I get an error about 'sowl' not being supported or something like that) Now I am trying quick time alternative in wine along with

I am playing with gstreamer on CentOS 7.1 . All the packages are there except gst-rtsp-server.... OK - So I went to look for it to compile it. Found a version that was NOT 1.0.7 and my compile stopped because it said the versions are not the same.... OK so went back to look for 1.0.7 and its not there. I looked at: http://gstreamer.freedesktop.org/src/gst-rtsp-server Any thoughts on how I

Atul S Vasu wrote: > Where do I find instruction to configure icecast as a reflect server for > rtsp? Icecast doesn't suppport RTSP. Only HTTP. Geoff.

hi All, I am using 11.25.2 and musing on hold. CentOS 7.4 I am trying to setup a MusicOnHold() streaming audio. I have one machine I tried this on and it worked. This machine is on the internet. I have another machine behind a firewall that does not play. Both machines installed mpg123. I have a windows machine behind the firewall that plays the audio stream so firewall is not the issue. I

Hi I need to stream ogg/theora over RTSP in order to use in opera browser HTML 5 video, but I cannot find any streaming server to do that. -- best regards, Daniel Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/theora/attachments/20100311/f4ec72d2/attachment.htm