similar to: Call Problems W/CVS Head

> it's fixed > > russell at lists.digium.com russell at lists.digium.com > Tue Jul 19 11:23:06 CDT 2005 > > a.. Previous message: [Asterisk-cvs] asterisk utils.c,1.58,1.59 > b.. Next message: [Asterisk-cvs] asterisk utils.c,1.59,1.60 > c.. Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] > >

Hi - I've just been testing out the latest CVS HEAD (as of about 10:00a EDT today). I'm getting some weird errors. Calls from one sip phone to another have OK audio in one direction and highly scrambled audio in the other direction. The console shows this error repeated ad nauseum during each call: Jul 18 16:08:03 ERROR[22941]: utils.c:509 tvfix: warning negative timestamp

i get lots of the below from friday 15.7.05 cvs as well ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

I have been trying to make Broadvoice match incoming Caller ID and do specific things based on the number received, but due to Broadvoice requiring the "s" to start off an incoming extension, I cannot get this to work. Has anyone been able to do this? Here are some examples of my setup: from sip.conf: context=broadvoice-incoming from extensions.conf [broadvoice-incoming] exten =>

Hello, I’ve a question. I have a station now with Shoutcast and shoutcast transcoder for 2 bitrates. 192kbps mp3 and 64 kbps aac+. I want to transfer these to Icecast. I’ve an working icecast server but i need an auto dj that can encode 2 bitrates, 192kbps mp3 and 64kbps aac. I had a working setup, but my nonstop machine is not on the icecast server but externally. Now in shoutcast transcoder i

On January 15, 2018 1:53 AM, Wayne Davison write: >I have made rsync 3.1.3pre1 available for testing. This release has a couple security fixes, a few new features, and a smattering of bug fixes. >Please test this new release and send email to the rsync mailing list with any questions, comments, or bug reports. Incompatibility introduced at 3.1.3pre1. These are not present at master, which

Greetings Rsync, Quick intro: I'm a port/platform maintainer for a variety of HPE NonStop Open Source ports including rsync, git, and a few others, and have been in the industry for far too long. Our port is currently at commit 1f83b51, which I'm planning to upgrade to 7706303 in the next few days. We had been able to stay on vanilla code until 1f83b51 when we had to introduce a

On Sun, Apr 26, 2020 at 11:32:10PM +0200, Marvin | OveralWiFi.nu wrote: > I’ve a question. I have a station now with Shoutcast and shoutcast > transcoder for 2 bitrates. 192kbps mp3 and 64 kbps aac+. > > I want to transfer these to Icecast. I’ve an working icecast server > but i need an auto dj that can encode 2 bitrates, 192kbps mp3 and > 64kbps aac. > > I had a working

On Sun, Sep 20, 2020 at 03:13:28PM -0400, Randall S. Becker wrote: > On September 20, 2020 2:02 AM, Damien Miller wrote: > > OpenSSH 8.4p1 is almost ready for release, so we would appreciate testing > > on as many platforms and systems as possible. This is a bugfix release. > > I will be testing this shortly on HPE NonStop platforms. > > Side question: We now have

https://bugzilla.samba.org/show_bug.cgi?id=13224 Bug ID: 13224 Summary: New file handling causes compile issues for NonStop port. Product: rsync Version: 3.1.3 Hardware: Other OS: Other Status: NEW Severity: normal Priority: P5 Component: core Assignee:

Greetings, I am porting the openssh-portable 6.8 release to the HP NonStop (NSE) platform. Prior versions were no real problem, with minor tweeks. However, with the inclusion of regress/netcat.c, which depends on arpa/telnet.h, we have an issue. Unfortunately, the platform does not have this file, nor anything like it - telnet is done rather differently. We do have a version of netcat (0.7.1

I'm seeing something unusual in 5.0p1. Let me start by saying that I'm on kind of an oddball system (HP NonStop). What I'm seeing is that at the end of an scp session, the server gets stuck in a loop. First I see a shutdown failure, followed by looping on an "ibuf_empty delayed efd 9/(0)" condition. This may have to do with some minor semantic differences in the way the

I currently have my astcc databases card lenghts at 7 digits long. I would like to expand this to 10 digits now though. Will I screw things up if I leave the old 7 digit long pins in there and start using/generating 10 digit pins? Thanks

I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week?

Probably doesn't help diagnose the problem.... but there were also audio problems experienced with this cvs version even on LAN / sip2sip / no transcoding > > ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... > > I will be looking into this issue later today. __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take

https://bugzilla.mindrot.org/show_bug.cgi?id=3205 Bug ID: 3205 Summary: Support HPE NonStop Server Port Product: Portable OpenSSH Version: 8.3p1 Hardware: Other OS: Other Status: NEW Severity: enhancement Priority: P5 Component: Build system Assignee: unassigned-bugs at

hello Since a while we're experiencing completely I/O stalls / stopps with some HP Blade Hardware and FreeBDS 9.1. If it occurs, we have to power cycle the server to recover. The stalls occur only every few days to weeks, so it's difficult to reproduce them; we also tried to do a lot of I/O (e.g. scrubbing nonstop) but this didn't help to reproduce more often; taking load away from

When I worked on the HPE NonStop compilers for x86 (we used Open64, not LLVM), we adjusted our -O1 to make sure the source display didn't "bounce around" based on feedback from users. We disabled any optimization that would move things across statement boundaries. We also disabled/de-tuned dead store since our DWARF location list support was pretty basic and with the removed store,

I was using the Asterisk::Manager perl module and had some troube with it so I decided to make my own. I have a pretty good prototype after a few hours (My main point about the protocol is at the bottom of this example.) http://asterisk.650dialup.com is where you can download it. my $man = init Asterisk::AstMan ({ -user => "asterisk",

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound