similar to: Campusing two Asterisk boxes?

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

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

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

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

Has anyone on here used kiax before? I am asking because I have it installed on several computers and have been able to get it to connect and register to my Asterisk box. I can even call between them and my SIP softphones. The problem I am having is this: when I use kiax to call someone else, they get some kind of background music playing while I am talking to them. I have called from kiax to

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

Hello everyone. I had heard about this open-source PBX once a while back. I wasn't too interested in it at the time but I kept the info filed away for possible future use. A couple of days ago, I was walking around Barnes and Nobles and I found this book, called Asterisk: The Future of Telephony. I paged through it a little and I was really excited by what I read. Then I remembered the

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 All, This is probably another one of those completely silly questions that I'm going to hit myself later on, but I have the simplest issue right now but I can't figure out why it's happening: I have a trunk from one * box (box a) to another * box (box b) the call comes in from box a with an extension 2222 which acc. to the peer config in sip.conf is set to use context

Basically, I am looking to set up an extension which will be used as a "help-line". I want it to function kind of like the bat phone from the old Batman series, where Commissioner Gordon would pick up the extension in his office and it would ring the phone back at Wayne's mansion. Is there a way to duplicate this functionality with Asterisk? I just need asterisk to auto-dial an

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.

I have asterisk running more or less ok but I would like to turn off call waiting and be selective about the incoming sip connections. This is running asterisk 1.2.8 with a fxs and fxo card and a configured voip (sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk. Problem 1) if someone is on the phone already and another call comes in for an already engaged extension I

I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten => s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the

Hi! I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with

I have the following setup: AnalogPhone1--TDM400P-ASTERISK---via SIP---Softswitch--------POTS Phone2 When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well. When making a call from Phone2, I get a dial tone but after dialing the number I hear nothing (no ringing tone). On Asterisk console it says that a call is coming in and that it is ringing Zap/2. I can also hear the

Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I

Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband,

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing