similar to: MeetMe not prompting for PIN

I would like to do something like: exten => s,1,playback(http://my.server.com/file.wav) I tested and it does not work. It seems highly likely that someone would already have done this one way or another. I know I could do a system wget and then play the local file, but wanted something a bit more elegant. Thanks, Mike Clark

I finally found a reason TO run Asterisk as root. By default, ext[23] file systems "reserve" 5% of the filesystem for root. Thus, you may get some warning when everything non-root starts failing and give you a chance to free up some space before Asterisk is affected. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards

I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, all. I've solved my own problem but will post it here in case someone else has the same misunderstanding in the future. We thought we had set up our meetme so that regular users entered the conference without a pin but could not speak to each other until the moderator arrived. We enforced pin entry on the moderator . . . or at least so we thought. If the moderator waited long enough

Dear Asterisk Users Mailing List - Non-Commercial Discussion, I joined VirtualPhoneLine.Com service and am really enjoying the use of it. VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, Google Talk or an IP Phone. Have a look at the http://www.virtualphoneline.com/faq and

hello is it possible to lock a conference IF no admin are connected ? or how to do to have a conference offline? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, How do you tell how much of your l2arc is populated? I''ve been looking for a while now, can''t seem to find it. Must be easy, as this blog entry shows it over time: http://blogs.sun.com/brendan/entry/l2arc_screenshots And follow up, can you tell how much of each data set is in the arc or l2arc? -- This message posted from opensolaris.org

Hi Everyone, I am using pfSense to do firewall and NAT on an Asterisk server. I have ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP 192.168.5.5. However, when a user from outside using Linksys WRP400 ata connects to the Asterisk server and registers I see them as 192.168.1.1 in the "sip show peers" command. In face, all many different of the Linksys WRP400

I''m sorry to have to bother the list with this, but I''m missing something obvious and my brain is cramping or something, I cannot find the answer... Here is an entry from my log file: Sep 16 11:12:34 fw kernel: Shorewall:rfc1918:DROP:IN=eth0 OUT= MAC=00:c0:9f:1e:fa:99:00:07:50:cd:a5:80:08:00 SRC=201.1.7.201 DST=208.10.57.129 LEN=48 TOS=0x00 PREC=0x00 TTL=111 ID=11155 DF

I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant High-Availability. I was wondering if it's possible for Asterisk to also use multiple database servers for Realtime? Currently with Realtime I am only able to point to a single IP address for a database. If that database server goes down that Asterisk is pointed to then Asterisk won't be able to do anything.

Hi, if anyone needs Gnu Gadu 2 [Instant Messaging Program] for Centos 4, I've compiled it (some packages from dag, possibly dries) and it's up at: http://tcs.ii.uj.edu.pl/~maze/centos/gnu-gadu-2/ Cheers, MaZe.

Hello, all. My apologies up front but I must be brain cramping on something very simple. I've tried to pare down my configuration to the absolute minimum for SIP traffic just to understand how it works. My incoming calls are not finding the "s" extension in my dial-plan. I am assuming SIP calls can do this. I am using Asterisk 1.6.1.1 sip.conf has nothing but: [general]

i tried using music on hold option in the dial command exten => 7777,1,Dial(zap/1/7777,20,m) when someone calls me and i picked up the phone, the call will be suddenly dropped. however, if i use a sip client instead of zap (also changing the dial statement to sip), i can answer the incoming call without a problem. is this a known bug? (asterisk cvs 05-03-04 using RedHat v9 on Via mini-ITX)

Hi again, I have improved precision of my codec (on the encoder and decoder).I have also improved the -h1 quality setting.-Source code and binaries at http://nhwcodec.blogspot.com/-. I am still trying to improve precision of my codec, with keeping my low-complexity (fast) approach. I do not totally use the reference (and impressive) block prediction with different modes + residual coding scheme,

Hello list, someone (actually neil perrin (CC)) mentioned in this thread: http://mail.opensolaris.org/pipermail/zfs-discuss/2009-December/034340.html that is should be possible to import a pool with failed log devices (with or without data loss ?). >/ />/ Has the following error no consequences? />/ />/ Bug ID 6538021 />/ Synopsis Need a way to force pool startup when

Hi, yesterday I was trying to create local mirror of CentOS7 using katello/pulp. I was receiving CRC checksum errors while attempting to sync the mirror. The particular error in question was: Feb 12 07:46:34 bazinga pulp: pulp_rpm.plugins.importers.yum.sync:INFO: [e2cd9930] Generating metadata databases. Feb 12 07:47:06 bazinga pulp: pulp_rpm.plugins.importers.yum.sync:ERROR: [e2cd9930]

I am using asterisk (2 week old CVS) am for the first time have been starting to experiment with busy and congestion. At this point I am only using sip endpoints PAP2-NA devices. All testing of this is being done on a local network. my test extension looks like this: exten => 7777,1,Answer exten => 7777,2,busy(35) exten => 7777,3,Hangup Or like this: exten => 7777,1,Answer

I found "7777" and "*65" are not working. Please how can i re-enable them again. Thanks _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.

Hi: Sad question, I know. I saw it somewhere but can''t rediscover it. I have a collection_select and want to add a prompt so the dropdown displays "select one" How is that done? thanks. bruce