similar to: limit-rate

Hello, Thank you for your response. It is on the kh version.. https://github.com/karlheyes/icecast-kh Le sam. 3 nov. 2018 à 21:47, Thomas B. Rücker <thomas at ruecker.fi> a écrit : > Hi, > > On 11/03/2018 07:33 PM, Mickael MONSIEUR wrote: > > Hi, > > Where is the mount option 'limit-rate' in the current version? > > I checked in cfgfile.c and in the

On 11/03/2018 09:53 PM, Mickael MONSIEUR wrote: > Hello, > Thank you for your response. > It is on the kh version.. That's not a version. That's completely different software at this point. It's also not Xiph.org, but published by Karl. TBR > Le sam. 3 nov. 2018 à 21:47, Thomas B. Rücker <thomas at ruecker.fi > <mailto:thomas at ruecker.fi>> a écrit :

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, When I get a SIP INVITE as follows: INVITE sip:s at 10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: "0475XXXXXX" <sip:1053212 at sip.domain.com>;tag=as7df9ab18 To: <sip:02XXXXXX at IP:5060> Contact: <sip:1053212 at IP:5060> Call-ID: 344d42bd16975a54141d11f635bdfc71 at sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL,

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have

Good morning, or Good afternoon! It depends :-) I have a standard Asterisk configuration: SIP friends (phones) <-----> Asterisk <-----> SIP gateway to PSTN converter 80.236.215.61 109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:xxxx at 80.236.215.61:64946;ob

Hello, I look ARI (Asterisk Recording Interface) the publisher site is closed... http://www.littlejohnconsulting.com/ari Thank you, Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/a8d923ae/attachment.htm

Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-0000004d == Spawn extension (dialin, 065939191, 2) exited non-zero on

What is the interest to supply binary of Asterisk, under debian for example, while to use MeetMe it is necessary to COMPILE Asterisk ??? :-)) Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100611/e4d749f6/attachment.htm

Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/111111, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, The MeetMe application refuses MusicOnHold personalized and skip always in the default! Have you any idea how to fix this? -- Executing [028883899 at default:1] Set("SIP/109.10.214.1-00000002", "CHANNEL(language)=fr") in new stack -- Executing [028883899 at default:2] Answer("SIP/109.10.214.1-00000002", "") in new stack -- Executing

Hello, Is it possible to connect two *callers* without going through a conference (meetme) ? Example: 06:50pm - User 1 call extension 600 and musiconhold / parked call .. 06:51pm - User 2 call extension 600 and connect to User 1. Thank you in advance, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have setup MRTG with 14all.cgi cgi script. I downloaded it from below URL. http://my14all.sourceforge.net/14all-1.1.txt and did only below 3 changes to that file. those can be seen in BOLDletters. # if MRTG_lib.pm (from mrtg) is not in the module search path (@INC) # uncomment the following line and change the path appropriatly: #use lib qw(/usr/local/mrtg-2/lib/mrtg2); use lib

Hello, I tried to build an debian package from svn. First I checked out trunk and the 2.3.2 branch, which both build this quite "old" package :-) icecast2_1.9+2.0alphasnap2+20030802-1_amd64.deb Then I tried the kh branch, which doesn't build with dpkg-buildpackage: debian/rules:6: debian/cdbs/1/rules/buildinfo.mk: No such file or directory make: *** No rule to make target

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

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

Say I have a list of variables,  listVar <- list(age,sex) I am looking for a way to either 1- create a vector c("age","sex") from it, or 2- get the names one by one in a for loop such as these     a)  for (i in 1:length(listVar)) rownames(result)[i] <- ???     b)  for(i in listVar) print (variable's name) Any help much appreciated. [[alternative HTML version

Hello, I didn't want to have to choose between Icecast running on port 80 and all my Apache virtual hosts, running also on port 80, on my sole external IP address. I didn't want either to open port 8000 on my firewall because I wanted all users being able to reach Icecast even the ones behind enterprise firewalls. So I managed to run Apache 2.2 listening on port 80 and Icecast 2.3.2

I've seen a number of issues with the loading of workspaces discussed previously, but here's another one... I simply can't load any saved workspace at all... Here's an example, starting with an empty workspace and creating a single variable "a". > a<-1:5 > save.image("a.Rdata") > rm(a) > load("a.Rdata") Error in function ()  : unused