search for: find_us

Displaying 20 results from an estimated 39 matches for "find_us".

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2003 Oct 20
3
Call Waiting on SIP phones
...if ( u->outUse > 0 ) { u->outUse--; @@ -994,6 +998,7 @@ } u->outUse++; break; + */ default: ast_log(LOG_ERROR, "find_user(%s,%d) called with no even t!\n",u->name,event); } @@ -1086,6 +1091,12 @@ INVITE, but do set an autodestruct just in ca se. */ needdestroy = 0; sip_scheddestroy(p, 15000); +...
2006 Jun 11
2
Finding a record and showing it -- how?
I''d like to prompt a user for the value of a Name field, then display the record. Rails tells me that it cannot do a find without an ID. I guess it must be that I''m not passing back properly the data from the view to the controller. Thanks for the help joshi The find_user.rhtml view: <div class="find-name-form"> <fieldset> <legend>Enter User Name</legend> <%= start_form_tag %> <p> <label for="name">Name:</label> <%= text_field_tag :name, params[:name] %>...
2006 Apr 19
4
Another DRY question
...lly add the user''s id to the user_id field in the product table. I don''t want to keep typing the user = User.find stuff over and over again so thought that I could create a private method and have it work that way, but I''m having trouble with.. syntax. private def find_user user = Usr.find(session[:user]) user_id = user.id end end And then changing my list to def list @user = find_user @product_pages, @products = paginate :products, :per_page => 10, :conditions =>[''user_id = ?'', @u...
2006 Apr 25
5
where does logic for the layout go?
on the main layout for my controller, i need a list of users - each with their own link to a lists of tasks that they have been assigned. i did have a line inside the .rhtml file that found all the users, but i''m guessing this is probably the wrong place to put them. i''m new to ruby and rails, so i still struggle a little with the MVC concept. -- Posted via
2004 Jun 23
0
UPDATE Patch for postgres enabled app_voicemail.c
...dboption); if (PQstatus(dbhandler) == CONNECTION_BAD) { *************** *** 253,270 **** } ast_mutex_init(&postgreslock); ! /* fprintf(stderr,"postgres login OK\n"); */ return(0); } ! static void sql_close(void) { ! PQfinish(dbhandler); } static struct ast_vm_user *find_user(struct ast_vm_user *ivm, char *context, char *mailbox) { PGresult *PGSQLres; --- 269,371 ---- } ast_mutex_init(&postgreslock); ! fprintf(stderr,"postgres login OK\n"); return(0); } ! ! static void sql_append_mailboxes(void) { ! /* ! * Matt Davies (aka JMD) ! * matt...
2004 Jun 23
0
Patch for postgres enabled app_voicemail.c
...dboption); if (PQstatus(dbhandler) == CONNECTION_BAD) { *************** *** 253,270 **** } ast_mutex_init(&postgreslock); ! /* fprintf(stderr,"postgres login OK\n"); */ return(0); } ! static void sql_close(void) { ! PQfinish(dbhandler); } static struct ast_vm_user *find_user(struct ast_vm_user *ivm, char *context, char *mailbox) { PGresult *PGSQLres; --- 269,371 ---- } ast_mutex_init(&postgreslock); ! fprintf(stderr,"postgres login OK\n"); return(0); } ! ! static void sql_append_mailboxes(void) { ! /* ! * Matt Davies (aka JMD) ! * matt...
2003 Oct 23
6
Problems with * and IAXTel/FWD
...at I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1 DEBUG[1133735216]: File chan_sip.c, Line 973 (find_user): Call from user 'phone1' is 1 out of 0 DEBUG[1133735216]: File chan_sip.c, Line 3307 (build_route): build_route: Contact hop: <sip:phone1@10.1.2.24:5060;line=1> -- Executing Dial("SIP/phone1-2c71", "IAX/user:secretpass/BYEXTENSION@iaxtel") in new stack -...
2003 Sep 03
8
Asterisk Jitters
...but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user 'xirak' is 1 out of 0 DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: Contact hop : <sip:192.168.7.3> -- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write):...
2006 May 22
2
good practice or waste of time?
...ing the acts_as_authenticated plugin I have setup the appropriate pieces and use a before_filter to require the session to be that of a logged in user. So far, so good. Here''s where things (finally) get interesting. When my controller needs to edit an item, I typically do this: def find_user @user = User.find_by_login(session[:user]) end def find_library find_user @library = @user.library end def edit_item find_library @item = @user.library.items.find(params[:id]) <--- necessary? # other processing end It''s that second line in my #edit_item action tha...
2006 Jul 27
1
response from ajax request
...t I am targeting, the plain html elements from the partial template have been stripped, leaving only the elements inserted via the <%= statements. I would appreciate any hints. Best, Eric here is the controller action that responds to the ajax request: =============================== def find_users @user_is_admin = is_admin @user_pages, @users = paginate :users, :order_by => ''login'', :per_page => 10, :conditions => "name like ''%kyle%'' or children like ''%kyle%''" # add error checking later...
2006 Jun 20
0
ooh323 issues
...to my Asterisk server, my SIP phone rings, and I get a ringing signal from the H.323 server, but when the SIP phone is answered, it goes dead with the following error message: **** --- onNewCallCreated ooh323c_10 +++ onNewCallCreated ooh323c_10 --- ooh323_onReceivedSetup ooh323c_10 --- find_user +++ find_user Adding capabilities to call(incoming, ooh323c_10) Adding gsm capability to call(incoming, ooh323c_10) Adding g711 ulaw capability to call(incoming, ooh323c_10) Adding g7231 capability to call (incoming, ooh323c_10) --- configure_local_rtp +++ co...
2003 Oct 13
1
out going calls
...a7-4d5e38ae@210.9.49.216' of Response 101: Found Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 4891 (handle_request): Check for res for mick Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 973 (find_user): Call from user 'mick' is 1 out of 0 Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '000af4bb-75e0182e-380c5ba7-4d5e38ae@210.9.49.216' of Response 102: Not Found Any ideas would be appreciated Regards Mick West
2003 Oct 21
1
Hangup
...EBUG[1605650]: File chan_zap.c, Line 1960 (zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/1-1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 992 (update_conf): Updated conferencing on 1, with 0 conference users Oct 17 19:20:02 DEBUG[1605650]: File chan_sip.c, Line 985 (sip_hangup): find_user(atasuporte) Oct 17 19:20:04 DEBUG[147466]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 ============================================ debug when I hangup the ATA186 =========================== Oct 17 19:40:25 DEBUG[278546]: File dsp.c, Line 1212 (ast_dsp_pro...
2003 Nov 06
0
SIP nat not working with budgetone (long)
...6@192.168.0.100' of Response 32119: Found Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c, Line 3908 (check_user): Setting NAT on RTP to -1 Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c, Line 4962 (handle_request): Check for res for 1747xxxxxx Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c, Line 985 (find_user): Call from user '1747xxxxxxx' is 1 out of 0 Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c, Line 3355 (build_route): build_route: Contact hop: <sip:1747xxxxxxx@192.168.0.100:51332> Nov 6 01:50:14 WARNING[4101]: File chan_sip.c, Line 456 (retrans_pkt): Maximum retries exceeded on cal...
2004 Jul 15
3
SIP to H323 call timeout
...12:1720, timeout=0. Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256, data=h@10.10.1.12:1720. Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter Jul 15 16:14:31 DEBUG[311316]: is not a local user Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on '842589597@10.10.1.13' of Response 1: Found CONSOLE Output ============== *CLI> -- Executing Dial("SIP/-08121388", "h323/1561...
2006 Apr 04
1
asterisk-ooh323, asterisk 1.2.6 and netmeeting
...ting calls an IVR-related app like Directory, SayDigits et al. my ooh323.conf file is attached. also, here's the asterisk console output with ooh323 debug on: NetMeeting H323 to SIP --- onNewCallCreated ooh323c_7 +++ onNewCallCreated ooh323c_7 --- ooh323_onReceivedSetup ooh323c_7 --- find_user +++ find_user Adding capabilities to call(incoming, ooh323c_7) --- configure_local_rtp +++ configure_local_rtp +++ ooh323_onReceivedSetup - Determined context default, extension 6384 --- onAlerting ooh323c_7 --- find_call +++ find_call +++ onAlerting ooh323c_7 -- Executi...
2004 Mar 05
3
dropped calls
...is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' ----------------- The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN...
2009 Jun 13
2
removing Mocha; 'spec spec' fails but the specific model file passes
I happened to mix ryan bates'' authentication scaffold with rspec_scaffold on a demo project. and ran into the problem of mixing mock frameworks...ryan uses mocha. So, as a learning experience, I choose to redo ryan''s tests without mocha but ran into a strange problem with tests of the User model. With debugging you can see.... If you run just the user_spec.rb file, everything
2008 Mar 27
1
Problem when leaving voicemail
...] DEBUG[19786] pbx.c: Launching 'VoiceMail' [Mar 26 16:25:03] VERBOSE[19786] logger.c: -- Executing [ 101 at enswitch-mailbox:3] VoiceMail("SIP/5060-ac017e30", "101 at 708|us<101 at 708%7Cus>") in new stack [Mar 26 16:25:03] DEBUG[19786] app_voicemail.c: Before find_user [Mar 26 16:25:03] DEBUG[19786] app_voicemail.c: In find_user_realtime for mailbox 101 context 708 [Mar 26 16:40:24] VERBOSE[14269] logger.c: Really destroying SIP dialog '1004787600 at 127.0.0.1' Method: OPTIONS [Mar 26 16:40:28] ERROR[19786] res_config_mysql.c: MySQL RealTime: Ping fail...
2003 Nov 17
1
ISDN debugging and SIP dial-in issue]
...407 Proxy Authentication Required [ ... ] ACK sip:2@192.168.1.10 SIP/2.0 [ ... ] DEBUG[5126]: File chan_sip.c, Line 565 (__sip_ack): Stopping retransmission on 'b5288d54-a46c-9e16-ff7c-ec43221a71b2@192.168.1.190' of Response 53320: Found [ ... ] DEBUG[5126]: File chan_sip.c, Line 991 (find_user): Call from user '17476691152' is 1 out of 0 Looking for 2 in localphones DEBUG[5126]: File chan_sip.c, Line 3369 (build_route): build_route: Contact hop: <sip:17476691152@192.168.1.190> -- Executing Playback("SIP/17476691152-a52e", "publicar-extbusy|skip")...