similar to: vorbisRTP and theoraRTP

Hi. Here it is my patch to put a video preview of a theora stream in status.xsl. I just added a: <video-preview>1</video-preview> parameters in icecast.xml.in that control the previewing function. It encodes a png in $webroot/$mountname.tmp and then move it to $webroot/$mountname.png As for now it saves a frame every theora keyframe, which is probably too heavy for the server but

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Updated version of video preview covering frame writing every 3 keyframe and a xsl typo. Best regards :) kysucix -- Make things as simple as possible, but no simpler. - Albert Einstein

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 <p><p>re all! about the running question (as i read on the http://icecast.org website in the 3rd party software section) if MuSE is working with icecast2, here is the answer: it definitely does :) the problem was fixed a couple of months ago in the CVS code, now we have a new stable release which will get hopefully soon properly packaged

Good Morning, After further investigation, I found that the res_pjsip_publish_asterisk module does not use the realtime sorcery wizard, but instead only reads from the configuration files. I've been able to patch the module, using the logic from the other modules to learn how to make the sorery configuration read from the other sorcery wizards and it's now working for the

Hello, Has anybody seen that Audiocodes gateway is replying with "486 Busy here" when it's actually not (last call ended ~15 seconds ago). I see this quite often. From other logs i see that previous call ends at 11:13:01, then app_queue tries to dial at 11:13:14 and fails numerous times, before succeeding at 11:14:02 I have attached sample SIP debug log: Any ideas what i could

hem here it is the patch. ;) bye kysucix -------------- next part -------------- Index: conf/icecast.xml.in =================================================================== --- conf/icecast.xml.in (revisione 10365) +++ conf/icecast.xml.in (copia locale) @@ -62,6 +62,7 @@ <port>8001</port> </listen-socket> --> +

On Fri, May 1, 2015 at 10:43 AM, Robert Broyles <robert at webservicesaz.com> wrote: > We love our Digium phones and DPMA - but we really need it to work on our > Realtime Platform. Otherwise we lose all the cool features and they are > just standard SIP phones. > > Anyone working on a solution for this? Or anyone from Digium see this on > the roadmap? > Hey Robert -

My database machine is broken and I have to use another one. I made somewhere mistake(s) and get now in the debug file: [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query: SELECT * FROM sip_buddies WHERE name = '886' [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query Failed because: Can't find file: './astconf/sip_buddies.frm' (errno: 13) [Feb 24 09:05:25]

Matt Hoskins wrote: > Good Morning, > > After further investigation, I found that the res_pjsip_publish_asterisk > module does not use the realtime sorcery wizard, but instead only reads > from the configuration files. I've been able to patch the module, using > the logic from the other modules to learn how to make the sorery > configuration read from the other sorcery

I try to get Realtime to work, ... the debug looks like below. Mar 12 00:56:56 DEBUG[25640]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '621' Mar 12 00:56:56 DEBUG[25640]: MySQL RealTime: Everything is fine. Mar 12 00:56:56 DEBUG[25640]: Unable to find key '621' in family 'SIP/Registry' Mar 12 00:56:56 DEBUG[25640]: Setting NAT on RTP to 524288

I am running asterisk 1.2.12.1 JK, Message: 26 Date: Thu, 24 May 2007 21:40:31 -0700 From: JK <jk@bingoconsulting.com> Subject: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000 To: asterisk-users@lists.digium.com Message-ID: <465668BF.6080800@bingoconsulting.com> Content-Type: text/plain; charset="iso-8859-1" Hello asterisk-users list. I

Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and have made it further, but am having a little difficulty. The outbound-publish object types seems to be working in realtime now. But the asterisk-publication object is only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't

On Thu, Feb 19, 2015 at 9:15 AM, Joshua Colp <jcolp at digium.com> wrote: > Matt Hoskins wrote: > >> Good Morning, >> >> After further investigation, I found that the res_pjsip_publish_asterisk >> module does not use the realtime sorcery wizard, but instead only reads >> from the configuration files. I've been able to patch the module, using >>

Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. In our scenario the SP is sending call to our ser server and ser is forwarding the call to asterisk. In the asterisk debug I can see the DTMF keys are coming but ivr does not recognice those keys at all. I can see this in the

My friends, I'm having some problems in my Asterisk, the thing is that Asterisk seem to be crashed (or dead) sometimes (2 times in 3 weeks) I noticed this today, when i could not make any internall call, tha calls to the voicemail (*1) did not work it just don't say nothing, nothing appears in console; i tried to make a CLI>stop now but nothing happens, i could not stop the asterisk

I`m making my radio station with Icecast. On Windows desktop everything worked perfectly but on my private Ubuntu-based laptop it`s not so easy... I` m using a tutorial here: http://idjc.sourceforge.net/tutorials_icecast.html When I try to run Jack server I get something like this: "Jack is running in realtime mode, but you`re not allowed to use realtime scheduling. Your system has an audio

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context |

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context | secret |