similar to: single sign on - with other apps?

I''m new to OmniAuth and am trying to get two sample apps (a provider and a client) to work. I have two sample apps I forked from a blog post on omniauth and single sign in: http://blog.joshsoftware.com/2010/12/16/multiple-applications-with-devise-omniauth-and-single-sign-on/ Provider - https://github.com/RobZolkos/sso-devise-omniauth-provider Client -

i am working on a site taht uses loginGenerator.. it''s a great little package, but i am running into one little problem.. i would like to have something that says: "logged in as <username>" in the site, but i am not quite getting it.. what i am using is: <p>You are logged in as: <%= @session[:user] %> but i have tried: <p>You are logged in as:

* compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin H323 debug enabled --

I cannot mount my Samba shares. I could, then I rebooted my device, and now I no longer can. The .conf file has not been changed in over a year. Here is the output of the log file: -bash-2.05b$ cat log.192.168.1.147 [2011/08/23 00:41:10, 0] lib/util_tdb.c:tdb_log(662) tdb(/opt/local/var/locks/sessionid.tdb): expand_file to 20480 failed (No space left on device) [2011/08/23 00:41:10, 1]

I issued the following curl command to post to the database using REST: curl -i -X POST -d "<heartrate><heartRate>102</ heartRate><sessionID>1002</sessionID><timeStamp>15068</timeStamp></ heartrate>" http://localhost:3000/heartrates I have tested the GET (with curl also) and it works fine! curl http://localhost:3000/heartrates/45

My configuration is: [ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] -- [CISCO ip phone 12SP+/Skinny] When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN everything working ok (RTP is ok). But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone IP phone party can hear ISDN party, but ISDN (incoming) party canNOT hear IP phone party

I have a factor called 'utm_medium' in the dataframe 'data' > str(data$utm_medium) Factor w/ 396925 levels "","affiliateID=&sessionID=0000821850667323ec6ae6cffd28f380&etag=",..: 366183 355880 357141 20908 357513 365348 368088 360827 31704 364767 ... The data in this factor is delimited with '&'. I basically want the affiliateID,

Try sending 5350 config and oh323.conf, versions, etc... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj. Sent: Wednesday, December 22, 2004 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party

Greetings, I've been subscribed to this list for some time and I'd like to start off by thanking everyone who helps out on it. This is my first post to it, so please be gentle :-) I have several offices set up with RedHat and CentOS terminal servers. We are using CentOS 4.6 and RedHat 4.6 on them, GDM for the display manager, and PXES (http://sourceforge.net/projects/pxes/) for the

Hi all, I am new to the list, I have setup samba using solaris package samba-3.0.2a I am running solaris 8. Everything seems to be working fine. But one thing, when I bash-2.03# ./smbstatus sessionid.tdb not initialised Service pid machine Connected at ------------------------------------------------------- No locked files bash-2.03# I got a "sessionid.tdb not

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am just curious, about the following: I've upgraded from 3.0.1pre3 to 3.0.1rc2 and found that running testparm gives me sessionid.tdb not initialised, and then outputs what is expected. Everything else seems to work as before. I'm using Mandrake 9.2, with SRPMS built by Buchan Milne rebuilt locally, but without changes from the

Hi, a partner, who exchanges voip traffic with my asterisk box, complains, that asterisk ignores hints about ports to use. Hints about ports to use, seem to be a feature of H323. (I'm not firm enough with H323 to verify this.) The remote party opens the media-in channel: remote-ip:port-A -> local-ip:port-B My local Asterisk-box uses the same channel for media-out: local-ip:port-B ->

Hello Everybody, I have two rails app. Both have login pages for user authentication. The two apps have links to one another. Now what i need is when a user logs in to one app, he should not be again asked for login when he clicks the link for second app. Please tell me how to go on? Thanks in advance, Ak -- Posted via http://www.ruby-forum.com/.

I've run into a situation where Samba is having problems coping. I have filesystems on a SunCluster that my Samba server nfs mounts and then shares to our Windows PC's. I've twice had a situation where if I have a failover of my cluster during production time, Samba begins to use up all swap space, the system load goes extremely high (it went to 98 when this happened the other day),

Hello... I'm having problems with H323/G729 setup. Below is the output of h.323 debug when making a call. I use a SIP phone connected to an * box in the same LAN. The * connects to a h323/g729 PSTN terminator through internet. Calls rings and are answered in the other side, but I get no sound at all nor the other side does (complete silence in both sides). I thought this would just happen

Hi, I have been trying to output my graphs to a file (jpeg, pdf, ps, it doesnt matter) but i cant seem to be able to get it to output. I tried a few things but none of them worked and am lost as what to do now. I am using the scatter3d function, and it prints out the graphs on tot he screen without any problems, but when it comes to writing them to a file i cant make it work. Is there any

Hello... I'm having problems with H323/G729 setup. Below is the output of h.323 debug when making a call. I use a SIP phone connected to an * box in the same LAN. The * connects to a h323/g729 PSTN terminator through internet. Calls rings and are answered in the other side, but I get no sound at all nor the other side does (complete silence in both sides). I thought this would just happen

Hello *, is the rtpip patch still valid for asterisk 1.6 (with some code changes, obviously)? https://issues.asterisk.org/view.php?id=8161 Or, in asterisk 1.6 there is an alternative to using it? This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13:47:49.000000000 +0200 +++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200 @@ -987,9 +987,6 @@

Bonjour, Mon serveur Samba 2.2.1a (sur Suse 7.3 PPC) a quelques soucis depuis 2/3 jours. Le dernier en date : les utilisateurs sont authentifi?s mais le lecteur ne se connecte pas correctement dans le script (si on ouvre le lecteur concern? il est vide). L'architecture : 2 serveur Samba (fichiers / impression) avec 30 utilisateurs depuis 1 an plusieurs serveurs NT4 dont le contr?leur

R-Help, I am trying to reduce at data set to rows where a specified value occurs in a specified value. Below is a screen capture of Rgui: > nsmalldata <-read.csv("c:\\DATA\\UNITY\\\PASS0_DOWNFADE\\nsmall.csv") > nsmalldata BOARDNUMBER SESSIONID MATRIXID ARRAYPOINT Temperature PS1 PS2 PS13 PS14 PS15 1 LB0DC 3043 7757513 1 -9999 -9999