Displaying 20 results from an estimated 39 matches for "find_user".
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 = ?'',
@use...
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@b...
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@b...
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): O...
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 that...
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
re...
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
+++ conf...
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_proce...
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 call...
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",
"h323/156130...
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
-- Executing...
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 B...
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
failed...
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") i...