similar to: Multicast codec

http://lists.digium.com/pipermail/asterisk-users/2003-March/008088.html "why is there such a delay in getting ipv6 rolled out when it solves all these problems ?" === There are many reasons... 1. Leasing Address Space from the I* society (small s...aka the Big Lie Society) is not desirable by all people... 2. The IPv6 Privacy Problem...that is especially important in the area of computer

running Version 3.6.23-14.el6_6 I'm having slow save times with folders names ~something.. ie. /www/~bin/content.txt. doing this via a windows share.. when I save file it has a moment (few seconds) of hesitation.. then saves. When I do /www/bin/content.txt. the file saves instantly. Is there anywhere I can change a setting to fix this? or is this a bug. I've looked in the logs

Just idly watching * in console mode and saw that someone from 50.49.54.102 tried to register with my *. whois gives:- OrgName: Internet Assigned Numbers Authority OrgID: IANA Address: 4676 Admiralty Way, Suite 330 City: Marina del Rey StateProv: CA PostalCode: 90292-6695 Country: US NetRange: 50.0.0.0 - 50.255.255.255 CIDR: 50.0.0.0/8 NetName: RESERVED-50

Is there any "easy" way to add a custom subscribe handler? I have a set of users with Polycom phones that attempt to Events that Asterisk/PJSIP doesn't recognize, "call-info" and "as-feature-event". It just generates a warning, but it got me wondering if I could add my own handlers for those that didn't actually do anything but simply responded with a 200 OK.

Currently, the TFTP configuration file discovery process will request a set of files based on the UUID, MAC address, lease IP address and eventually fall back to "default": /mybootdir/pxelinux.cfg/b8945908-d6a6-41a9-611d-74a6ab80b83d /mybootdir/pxelinux.cfg/01-88-99-aa-bb-cc-dd /mybootdir/pxelinux.cfg/C000025B [...] /mybootdir/pxelinux.cfg/C /mybootdir/pxelinux.cfg/default

Hi all, I get "Unknown RTP codec 72 received" message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console

so i just found that one of my hosts is GENERATING these probe pairs, maybe every minute or two (note the sequence numbers): seq my host victim(s) --- ---------------- --------------- 24) 192.168.0.2:1121 <--> 216.52.3.2:2703 25) 192.168.0.2:1122 <--> 216.52.3.4:2703 39) 192.168.0.2:1124 <-->

After reading this article: http://www.theregister.co.uk/2005/08/31/blocking_chinese_ip_addresses/ I got to thinking that there is really no reason for *any* traffic to hit my servers that comes from anywhere outside North America. So I wrote the perl script at the end of this posting to extract selected IP ranges posted at iana.org and convert them into iptables rules blocking any traffic

Hi, I have an existing (Open)LDAP which I'm using with dovecot and I would like to implement quotas. I have global quotas working, but I would like per user quotas similar to what is described in the wiki (http://wiki2.dovecot.org/Quota/Configuration#LDAP) My problem is that there is no obvious attribute in the schemas shipped in the RHEL/Centos 6 to hold the quota string. Is it possible to

HI, is there are official MIME type for Vorbis.... audio/x-vorbis ? Dave --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

Martin Leese wrote: > Subject: [opus] MIME Types and File Extensions > To: opus at xiph.org > Message-ID: > <CAAzqGd_uzR646Nsdt=O2HDxLOYE2=K=5n9UOHLr3Y4BGzdVasw at mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > Hi All, > > On the Xiph Wiki page at: > https://wiki.xiph.org/MIME_Types_and_File_Extensions ... > Could somebody more

http://shorewall.net/pub/shorewall/2.2-Beta/shorewall-2.2.0-RC1 ftp://shorewall.net/pub/shorewall/2.2-Beta/shorewall-2.2.0-RC1 Problems Corrected: 1. The syntax of the add and delete command has been clarified in the help summary produced by /sbin/shorewall. New Features: 1. TCP OpenVPN tunnels are now supported using the ''openvpn'' tunnel type. OpenVPN

Or just set up sane firewall rules to allow the two to talk, but block external traffic. I have run this way for years - all add'rs ar IANA in a subnet block, and just that block is open locally, and all other external IP's severely restricted, and NUT works great . . . - Tim On September 12, 2016 11:47:28 AM CDT, "Stuart D. Gathman" <stuart at gathman.org> wrote: >On

Hi All, On the Xiph Wiki page at: https://wiki.xiph.org/MIME_Types_and_File_Extensions I added Opus to several of the sections, but wasn't sure whether to add "Theora + Opus" to: .ogv - video/ogg .axv - video/annodex As far as I can tell, according to the Opus FAQ at: https://wiki.xiph.org/OpusFAQ#Will_Opus_replace_Vorbis_in_video_files.3F Theora + Opus is

The IANA has pre-allocated id 4 for ed25519. If waiting on the IANA were a reason to delay applying the SSHFP_KEY_ED25519 patch, it needn't be any longer. I've proposed un-reserving hash type 0 to be a "NULL hash", for those who'd rather publish the public key unhashed. Even if zero for unhashed fails to gain traction, I hope to see something allocated for that. But

Hello, We (Google) have been experimenting with configuration and adjustments to CELT-only Opus that give good results for compressing ambisonic audio signals [1]. Based on our results so far, we would like to use Opus to encode spatial audio. We hope to make it easy/possible to use libopus with other common tools and software modules (ffmpeg/libav in particular). Based on my reading of the

Hi, below is a patch to contrib/cygwin/ssh-host-config and the corresponding README in the same dir. It adds a `--port' option to the config script to allow setting another port than 22 for sshd. Additionally the script used to add `sshd 22/tcp' to the services file while the IANA proposes `ssh 22/tcp' and `ssh 22/udp' as services entries. The new version removes old `sshd'

What is the default multicast codec for multicast in Asterisk 13 ? G.729 or G.711 or other ? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200401/b83fc8ca/attachment.html>

On 21/08/17 10:37, Gedalya wrote: > On 08/21/2017 07:28 AM, voytek at sbt.net.au wrote: >> is there a 'preferred way'? should I tell users to use 143 over 993 ? or >> 993 over 143? or? > There is no concrete answer. There are various opinions and feelings about this. > The opinion againt 993/995 is that these are not standard ports, Out of curiosity, is there a

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about