similar to: v1.8.4: Extension Not found in Context?

We have a load balancer sending requests to one of X boxes and one of N mongrel processes on that box. Since each mongrel processes is multi-threaded but it has a mutex around the section that calls rails, we end up with several requests queued up waiting when they could have gone to another box with a free process. For example, boxA, and boxB. boxA has mongrels 1 through 10 boxB has

Hey guys sorry for this newbie type post but I'm not getting it really. I see other posts that r asking a simliar question but I'm not understanding. Currently I have working boxA and BoxC. SSH is the only thing accepted. So currenty I'm backing up files on boxC with "rsync -e ssh user@publicIPforB:/home/backup/back_me_up /home/backup". Now I have added boxB in front of

Hi, I have searched the archived, and see this question has come up with some frequency, but the solution by the original poster is never posted or the question is not answered completely. I have also RTFM (rsync and ssh), and I've been trying examples that I have googled. So here it is again. BoxA --internet-- BoxB (accessable only by ssh) -- BoxC I want to backup BoxC to BoxA. I want

I have a simple requirement/test I'm trying to perform, but having difficulty. I have a system with 2 interfaces, BoxA: eth0? 172.26.50.102 eth1? 192.101.77.62 My goal is to have a tcp port built on BoxA such that hosts on the 192.101.77.0/24 network can reach a port on a different box on the 172.26.0.0/16 network. The target system is 172.26.10.120?? tcp/22 The port I wish to build is

The setup... Transport is via SSH. BoxA : Ubuntu 8.10 BoxB : Mac OS X 10.4.11 If I use rsync v2 to transfer data from BoxA to BoxB everything works fine. However, if I use rsync v3 to transfer the data, the following error is reported : rsync: mkstemp "/path/to/somefile/file.pm.JKQczN" failed: No such file or directory (2) If at all possible,

hello list, i need to setup an asterisk system with 5 ISDN trunks. i found C4 cards but they are very expensive. i found that if i use 5 AVM Fritz! cards it would be very cheap. i want to use 2 boxes. 3 in boxA +2 in boxB =5 isdn. and i want, this two boxs to work as a single box so that one box can share ISDN hardware from other box. this system will be serving a call center. currenly we are

Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608)

I have been looking around some and I can't seem to find anything which will answer my question. If I have two Asterisk boxes in different locations which are linked to each other over the internet, can I configure the boxes to use each other's lines as local? In other words, let's say Site A has Phone1 for a 1FB line going into it on an FXO port. Site B has Phone2 for a 1FB line

Quick and dirty too hungover to think/search... ;) Sorry list AsteriskBoxA -> SBC/PSTN voodoo -> world wide Interweb -> AsteriskBoxB Account on BoxA UsernameJohn<12125551212> Account on BoxB UsernameJohn<102> They're both the same users, had to do some funky trunking (managed firewall provider is playing not in my backyard games)... So anyway, I need to take specific

Hi list. We have 2 NAS-Devices, each mounted on the same (virtual) box. For our backup we use bacula. Bacula ist writing its files to the first NAS (192.168.1.9). These are big files, up to 160 GB. For implementing a good backup strategy we decided to mirror (sync) another NAS-Device (192.168.1.8) to have redundant bacula backups. If sync is done the Device will be kept at another location too.

Hi, I am just migrating to Centos from fedora core 3. I have 3 boxes involved here: Box A --Fedora 3 (doing the dns, http, sendmail serving duties) Box B --CentOS Box C --CentOS I used to run root passwordless ssh both ways from boxes A and B mainly set up to keep the 2 boxes in sync w/rsync ... these 2 boxes normally ran one being the slave and the other being the master...the master did all

I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. "dialplan show" gives me that the context is present: [ Context 'smvoice-mediaport' created by

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]

hi guys I'm trying a typical, I'd presume, setup: two libvirt hosts with one lxc guest each. boxA: $  ovs-vsctl show 1a6a2e4f-11f3-48df-becd-03b59d00a1fb     Bridge "ovsbr0"         Port nm-team             trunks: [55]             Interface nm-team         Port "vnet12"             trunks: [55]             Interface "vnet12"         Port

hello, i've rsync installed: rsync version 2.6.9pre3 protocol version 29 Capabilities: 64-bit files, socketpairs, hard links, symlinks, batchfiles, inplace, no IPv6, 32-bit system inums, 64-bit internal inums i'm PULLing via rsync+ssh from BoxA to BoxB, eg: root@localhost /tmp > /usr/local/rsync/bin/rsync \ --rsh=/usr/local/openssh/bin/ssh \

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit