search for: gotos

On Sat, 2020-10-17 at 09:09 -0700, trix at redhat.com wrote: > From: Tom Rix <trix at redhat.com> > > This is a upcoming change to clean up a new warning treewide. > I am wondering if the change could be one mega patch (see below) or > normal patch per file about 100 patches or somewhere half way by collecting > early acks. > > clang has a number of useful, new

I have set up an auto attendant and all is working but I am bothered by a long pause when switching between menus. This pause is between 5 and 7 seconds and is quite annoying. Is there anyway to address this. One other thing I find interesting is that when I move from the main menu to the sub menu the delay is there but when I move from the sub menu to the main menu the delay is not there.

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem

Add xfs_info to show the geometry of the xfs filesystem. Signed-off-by: Wanlong Gao <gaowanlong at cn.fujitsu.com> --- Hi Rich, I got an odd error, can you help me with this error or give me a debug method? Thanks, Wanlong Gao daemon/Makefile.am | 1 + daemon/xfs.c | 278 +++++++++++++++++++++++++++++++ generator/generator_actions.ml

OpenSSH Developers, I'm researching whether or not OpenSSH is a viable commercial alternative to F-SECURE SSH or SSH.COM's ssh, but I'm not getting the kind of results that I expected from a "Non patent encumbered ssh client". When I attempt to build OpenSSH against an OpenSSL build without rc5, idea, or rsa it bombs since OpenSSL doesn't place the header files in the

I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running A@H 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

I've not seen an answer to this in any forum. I make a call through Asterisk, with a VOIP phone, doesn't matter which. The call gets made, I leave a voicemail, or complete the call in some manner, and the other side hangs up. I hear a busy signal on the phone on my end. If I have an extension that looks like this, after the hangup() is executed, my phone gives busy signals until I

From: Matt Fleming <matt at console-pimps.org> Date sent: Wed, 10 Oct 2012 10:32:07 +0100 Subject: Re: [syslinux] Menu system bug - MENU DEFAULT not working > On Tue, 2012-10-09 at 23:09 +0200, Tomas M wrote: > > > Where xxx is a *different* menu. I count 4. Are you able to reproduce > > > the problem with a simpler config file? > >

I''m going to be removing hole extents in the near future so I wanted to make a sanity test for btrfs_get_extent to make sure I don''t break anything in the meantime. This patch just puts btrfs_get_extent through its paces by giving it a completely unreasonable mapping to look at and make sure it is giving us back maps that make sense. Thanks, Signed-off-by: Josef Bacik

> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Goto > > show application goto > > -= Info about application 'Goto' =- > > [Synopsis] > Jump to a particular priority, extension, or context > > [Description] > Goto([[context|]extension|]priority): This application will cause the > calling channel to continue dialplan execution at the

In preperation for getting rid of the osd layer; change the code to use native wait interfaces. As part of this, fixed the buggy implementation in the osd_wait_primitive where the condition was cleared potentially after the condition was signalled. Signed-off-by: K. Y. Srinivasan <kys at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> ---

In preperation for getting rid of the osd layer; change the code to use native wait interfaces. As part of this, fixed the buggy implementation in the osd_wait_primitive where the condition was cleared potentially after the condition was signalled. Signed-off-by: K. Y. Srinivasan <kys at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> ---

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,

On Fri, Jun 28, 2019 at 3:35 PM James Y Knight <jyknight at google.com> wrote: > On Fri, Jun 28, 2019 at 5:53 PM Bill Wendling <isanbard at gmail.com> wrote: > >> On Fri, Jun 28, 2019 at 1:48 PM James Y Knight <jyknight at google.com> >> wrote: >> >>> On Fri, Jun 28, 2019 at 3:00 PM Bill Wendling <isanbard at gmail.com> >>> wrote:

On Mon, Jul 1, 2019 at 6:25 PM Finkel, Hal J. <hfinkel at anl.gov> wrote: > On 7/1/19 1:38 PM, Bill Wendling via llvm-dev wrote: > > On Fri, Jun 28, 2019 at 3:35 PM James Y Knight <jyknight at google.com> > wrote: > >> On Fri, Jun 28, 2019 at 5:53 PM Bill Wendling <isanbard at gmail.com> wrote: >> >>> On Fri, Jun 28, 2019 at 1:48 PM James Y

? 2023/3/28 20:04, Xuan Zhuo ??: > virtnet_build_xdp_buff_mrg() and virtnet_xdp_handler() auto I think you meant virtnet_xdp_handler() actually? > release xdp shinfo then the caller no need to careful the xdp shinfo. > > Signed-off-by: Xuan Zhuo <xuanzhuo at linux.alibaba.com> > --- > drivers/net/virtio_net.c | 29 +++++++++++++++++------------ > 1 file changed,

Hi, I was trying to use 2 of asterisk servers and interconnected, one of them as a peer to other sever (configured in sip.conf), so all the calls to server 1 will just be passed to server 2 (has PRI Card, TE 412P, only one PRI connected), i was sending calls to server 1 and that would send to server 2 and then dial out using Dahdi, but the problem that i got was the hangup cause codes, i was not

virtnet_build_xdp_buff_mrg() and virtnet_xdp_handler() auto release xdp shinfo then the caller no need to careful the xdp shinfo. Signed-off-by: Xuan Zhuo <xuanzhuo at linux.alibaba.com> --- drivers/net/virtio_net.c | 29 +++++++++++++++++------------ 1 file changed, 17 insertions(+), 12 deletions(-) diff --git a/drivers/net/virtio_net.c b/drivers/net/virtio_net.c index

...ot; if the goto frees "buffer". Avoid -using GW-BASIC names like "err1:" and "err2:". Also don't name them after the -goto location like "err_kmalloc_failed:" +using GW-BASIC names like "err1:" and "err2:". The rationale for using gotos is: @@ -440,6 +439,47 @@ A common type of bug to be aware of is "one err bugs" which look like this: The bug in this code is that on some exit paths "foo" is NULL. Normally the fix for this is to split it up into two error labels "err_bar:" and "err_foo:&quot...