Displaying 16 results from an estimated 16 matches for "ecreate".
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2019 Jan 25
0
[klibc:update-dash] redir: Fix typo in noclobber code
...), 1 deletion(-)
diff --git a/usr/dash/redir.c b/usr/dash/redir.c
index f96a76bc..71b0f774 100644
--- a/usr/dash/redir.c
+++ b/usr/dash/redir.c
@@ -192,7 +192,7 @@ openredirect(union node *redir)
} else if (!S_ISREG(sb.st_mode)) {
if ((f = open64(fname, O_WRONLY, 0666)) < 0)
goto ecreate;
- if (fstat64(f, &sb) < 0 && S_ISREG(sb.st_mode)) {
+ if (!fstat64(f, &sb) && S_ISREG(sb.st_mode)) {
close(f);
errno = EEXIST;
goto ecreate;
2020 Mar 28
0
[klibc:update-dash] dash: redir: Fix typo in noclobber code
...), 1 deletion(-)
diff --git a/usr/dash/redir.c b/usr/dash/redir.c
index f96a76bc..71b0f774 100644
--- a/usr/dash/redir.c
+++ b/usr/dash/redir.c
@@ -192,7 +192,7 @@ openredirect(union node *redir)
} else if (!S_ISREG(sb.st_mode)) {
if ((f = open64(fname, O_WRONLY, 0666)) < 0)
goto ecreate;
- if (fstat64(f, &sb) < 0 && S_ISREG(sb.st_mode)) {
+ if (!fstat64(f, &sb) && S_ISREG(sb.st_mode)) {
close(f);
errno = EEXIST;
goto ecreate;
2010 Feb 02
2
Semi-Transfer
There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing
that. Below is what I'm using in my extensions.conf
exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5)
exten=> X,2,SayDigits(${num})
exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify)
exten=>
2009 Apr 23
3
AGI PHP script
I have the below script that doesn't seem to be working. I don't know if
I have something in the script wrong that I am just missing. Or if I
don't have the php.ini set correctly for emailing
This is the CLI output
-- Executing [4099XXXXXX at port3_real:1] Goto("DAHDI/50-1", "newhire,s,1")
in
new stack
-- Goto (newhire,s,1)
-- Executing [s at
2009 Dec 02
2
Variable Name needed
Other than having stripping out IPs this is what I am receiving for my
voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works
fine calls that come in on my PRI. BUT at least from this VOIP source
the To field which is my RDNIS information for these calls, doesn't
actually fill into ${CALLERID(rdnis). But as you can see I'm getting the
information.
My question is, Does
2006 Mar 29
2
save related models from one form
I have three tables customers, indentities and people. I hav ecreated
models for them. The are connected with foreign keys and
has_many/belongs_to in this fashion:
customers has_many: identities
identities belongs_to: customer
identities has_many: people
people belongs_to: identities
Now, I have a form in which I would like to create db entries for the
three table...
2009 Aug 21
1
Queue Question
First off this is not my work for extensions.conf it is modified from
http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl
ogin-to-standard-dialplan-methods-part-1/
So credit to Leif Madsen <http://www.leifmadsen.com>
But as to my question
[AgentLogin]
;A replaced version of AgentCallbackLogin() using a GoSub()
;
exten =>
2009 Dec 04
1
IAX2 Port issue
Trying to configure IAX for use
I think I have everything set right. But my IAX phone wont connect.
When I run wireshark I'm seeing this
Note if above screenshot from wireshark does not show here is a link for
it: http://img402.imageshack.us/i/tempe.jpg/
I've tried a variety of setups in my IAX.conf (they all end up with the
same issue, tried just bindaddr=0.0.0.0 with
2009 Dec 02
0
FW: Variable Name needed
It might be worth mentioning the voip call is coming from a number we
have thru bandwidth.com in case anyone uses them.
James Shigley
From: James A. Shigley
Sent: Wednesday, December 02, 2009 3:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Variable Name needed
That wasn't it either. I tried a few other likely fields from
2009 Jan 16
0
No subject
AGI is executable.
=20
Then run 'agi debug' from the asterisk cli, place a call and see what
was send and receive from your agi
=20
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A.
Shigley
Sent: April-23-09 12:26 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] AGI PHP script
=20
I have the
2009 Apr 10
2
IVR Survey
Alright I know how to do basic IVR in *. But what I'm working on trying
to do now is a survey. I've found very little things out there on google
or the archives for how to do surveys with the * ivr.
Here is more or less what I'm trying to accomplish
1. Call comes in Plays Greeting
2. Starts Survey
3. Ask Q1, Record the answer (voice responses) repeat this
2009 Dec 28
2
SIP Issue
Alright I have a SIP phone located off premises with a very annoying
issue.
Well I say a sip phone it is actually two phones hooked to a Cisco Spa
2102
Link: http://www.cisco.com/en/US/products/ps10026/index.html
Each phone being a different line/extension.
Alright either line can ALWAYS make outbound calls no issue. The problem
is on the Inbound side. I'm completely stumped as
2009 Jan 16
0
No subject
is executable.
Then run 'agi debug' from the asterisk cli, place a call and see what was
send and receive from your agi
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A.
Shigley
Sent: April-23-09 12:26 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] AGI PHP script
I have the below
2010 Jan 05
6
Really Silly Question From Total Newbie
Hello All -
I've been poking around the past few weeks, trying to familiarize
myself with all of this. I am new to Linux, VoIP and Asterisk -- to
be complete. This is my first exposure to all of these technologies.
I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge
2400 and the install went well. I can log in and poke around in
Linux and I even configured the box to be
2009 Apr 13
10
Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
These are my settings :
sip.conf :
[root at asterisk asterisk]# cat
2007 Aug 23
0
[git patch] klibc dash 0.5.4 update
...FD2 && sv && sv->renamed[2] >= 0)
+ if (flags & REDIR_SAVEFD2 && sv->renamed[2] >= 0)
preverrout.fd = sv->renamed[2];
}
@@ -215,15 +214,17 @@ openredirect(union node *redir)
if ((f = open64(fname, O_WRONLY|O_CREAT|O_APPEND, 0666)) < 0)
goto ecreate;
break;
+ case NTOFD:
+ case NFROMFD:
+ f = redir->ndup.dupfd;
+ if (f == redir->nfile.fd)
+ f = -2;
+ break;
default:
#ifdef DEBUG
abort();
#endif
/* Fall through to eliminate warning. */
- case NTOFD:
- case NFROMFD:
- f = redir->ndup.dupfd;
- break;
case NHERE:
c...