search for: my_secret

All, I''ve upgraded to 2.0.2, and I can''t get my login screen (the first POST request in the application) to work. When I post this form, I see the "InvalidAuthenticityToken" error. I have protect_from_forgery :secret => ''my_secret'' set in application.rb and I am using an active_record session store based on this line in environment.rb: config.action_controller.session_store = :active_record_store My login_form is generated using form_for(). However, I am using text_field_tag and password_field_tag to generate...

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

...usr/lib/asterisk/modules/res_config_mysql.so exists. I have configured the following (after building a new DB with the appropriate SQL examples, with mods to drop the invalid keys, on the Wiki): - /etc/asterisk/res_mysql.conf [general] dbhost = 127.0.0.1 dbname = my_db dbuser = my_uname dbpass = my_secret dbport = 3306 dbsock = /tmp/mysql.sock - /etc/asterisk/extconfig.conf ; Extconfig.conf for realtime configuration voicemail => mysql,my_db,voicemail_users (Just want to try something simple such as voicemail for the initial testing.) I have removed the [default] section from my voicemail.co...

...d answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0<:1000@10.0.0.101> my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid="PSTN GW" <3000> deny=0.0.0.0 permit=10.0.0.154 ;SPA-3000 IP address dtmfmode=rfc2833 canreinvite=no cantransfer=yes My extension.conf [globals] PSTN_GW=10.0.0.154:5062 [from_pstn] exten => 1000,1,Goto(demo,s,1) [demo] exten => s,1,Ans...

...riend > ; username=testSIPtrunk > ; secret=test > ; host=10.10.10.16 > ; dtmfmode=inband > ; qualify=1000 Here's the beginning of the sip-vicidial.conf file: > ; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST > register => my_username:my_secret:my_username at myvoipprovider > > ; VICIDIAL Carrier: My VOIP provider > [myvoipprovider] > type=peer > host=adress.of.my.sip.server > fromuser=my_username > username=my_secret > secret=my_secret > context=default > dtmfmode=rfc2833 > disallow=all > allow=alaw &...

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

...exten => _7x.,4,Congestion [from-sip] exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => _7X.,1,Dial,Zap/1/${EXTEN:1} [pstn_fwd_forwarding] ;exten => s,1,Dial(IAX2/my_uname:my_secret@iax2.fwdnet.net/my_fwd_number|60|tT) exten => s,1,,BackGround(welcome) exten => s,2,Hangup [sip] include=IAXFWD include=IAXFWD-IN is there any suggestion for configuration. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digi...

...usr/lib/asterisk/modules/res_config_mysql.so exists. I have configured the following (after building a new DB with the appropriate SQL examples, with mods to drop the invalid keys, on the Wiki): - /etc/asterisk/res_mysql.conf [general] dbhost = 127.0.0.1 dbname = my_db dbuser = my_uname dbpass = my_secret dbport = 3306 dbsock = /tmp/mysql.sock - /etc/asterisk/extconfig.conf ; Extconfig.conf for realtime configuration voicemail => mysql,my_db,voicemail_users (Just want to try something simple such as voicemail for the initial testing.) I have removed the [default] section from my voicemail.co...

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP