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I have successfully installed IceCast with SSL support : https://relay2.dgnet.be/ Stream is working fine with this URL : https://relay2.dgnet.be:443/different.ogg Nevertheless the listen URL still points to http://relay2.dgnet.be:443/different.ogg ( without https then ) So the links provided/generated by IceCast are wrong . What did I miss ? Is it possible to force the listenurl value in the

Hello, Could you please explain or send an example for decoding a packet that contains multiple Speex frames. According to the API document the speex_decode_int() decodes only one frame of speech. Best regards, Danny Peled R&D Software Engineer Intelligence Solutions NICE Systems Ltd. ISRAEL *+ (972) 9 769 7161 ? + (972) 54 4 602840 *Danny.Peled at nice.com <mailto:*Danny.Peled at

Hi Justin, >Are you talking about the listen URL in the Icecast status page? I think it is currently not able show the link as httpS. Maybe one of the Icecast experts can confirm that. Yes I was talking about this value. The workaround was to put my own m3u file ( with the correct link ) in the icecast web folder > Is this truly an OGG audio file? Why does the server response give a

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

I have 2 drives in a Linux 7.1 server , 1 primary and 1 secondary. I am trying to clone to the secondary slave drive and be take the primary out and change the slave to primary and boot from it. The problem I am having is that it the copied drive will not boot, It is not getting the lilo boot and config properly. When I try to boot the copied drive , it boots to a Blank screen with an

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

Hello Asterisk users, I'm having a little problem with an Asterisk installation on Ubuntu, i had installed many asterisks on CentOS but never in Ubuntu, the problem is that Asterisk and DAHDI does not start at system start...i have to make "/etc/init.d/asterisk start" and "/etc/init.d/dahdi start" manually every time i reboot the machine (my laptop for testing) So, what

Hi, on 2 different servers, running 7.1-stable + zfs, I get this error rather frequently: Feb 5 17:01:03 warhol-00 kernel: impossible packet length (543383918) from nfs server sunfire:/dist Feb 5 17:01:03 warhol-00 kernel: impossible packet length (1936028704) from nfs server sunfire:/dist Feb 5 17:01:03 warhol-00 kernel: impossible packet length (1869363744) from nfs server sunfire:/dist

Hi, In my test case below, it seems that ''initialize'' is called everytime ''hello'' is called. Is this by design, a bug, or my pilot error? class FooController < ApplicationController def initialize print "\n initialize \n" end def hello print "\n hello \n" end end I

http://bugs.freedesktop.org/show_bug.cgi?id=19356 Summary: DPMS does not turn off LCD backlight on NV34M [GeForce FX Go5200] Product: xorg Version: unspecified Platform: PowerPC OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau

http://bugs.freedesktop.org/show_bug.cgi?id=16624 Summary: invalid pointer past to RRChangeOutputProperty Product: xorg Version: git Platform: All OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy:

Hi, between March 16 and now, bge on a Sun X2200 stopped working, turning off msi (via hw..pci.enable_msi=0) got it working again. I tried first replacing bge with an older version but that did not help. please advice :-) Danny

> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >

http://bugs.freedesktop.org/show_bug.cgi?id=19203 Summary: X default modes no longer added Product: xorg Version: git Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy: moondrake at

Many of you may have seen the recent announcement about Danny Windham coming on as the new CEO of Digium. This is one of the most exciting things to happen to Digium and to Asterisk at large. When Danny comes on board, I will be transitioning to the role of Chief Technical Officer (retaining my position of chairman of the board of directors), providing strategic vision for the company as

Hi All, I created a CLI tool that converts llvm coverage report JSON to a single HTML file that can be viewed locally. The tool is distributed from npm here https://www.npmjs.com/package/llvm-coverage-viewer . The code is hosted on github here https://github.com/hooddanielc/llvm-coverage-viewer. For any questions or concerns, please let me know or make an issue on github page. Hope this helps,

Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. In the extension.conf: -------------------------- [number-to-voice] exten => 8765,1,playback(num:344345,say) exten => 8765,n,hangup It executes corresponding say.conf script and produces good results for me. but when I write it in agi does not working. Here is agi debug output from asterisk.

No, this is a relay of a Shoutcast server. Everything is configured in https on IceCast ( https://relay2.dgnet.be/ ) , except this value listenurl ( in Admin / Mountpoint) which in cannot change apparently. So the .m3u file is created with an http:// link ( instead of https:// ) But the stream is available and working in https://relay2.dgnet.be:443/different.ogg Danny From: Icecast

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username