similar to: Invalid seqno and short packets

Dear Guus, I've attached my first git commit to your repository. It does not contain any new functionalities, but it is a first try to interact with your git copy. Could you please verify, if you can push this commit to your repository? If it works, I'll send you the rest of my work, which contains: 1) some small improvements in logging (using flags instead of counters) 2) the

From: Chia-I Wu <olvaffe at gmail.com> [ Upstream commit efe2bf965522bf0796d413b47a2abbf81d471d6f ] This is motivated by having meaningful ftrace events, but it also fixes use cases where dma_fence_is_later is called, such as in sync_file_merge. In other drivers, fence creation and cmdbuf submission normally happen atomically, mutex_lock(); fence = dma_fence_create(...,

From: Chia-I Wu <olvaffe at gmail.com> [ Upstream commit efe2bf965522bf0796d413b47a2abbf81d471d6f ] This is motivated by having meaningful ftrace events, but it also fixes use cases where dma_fence_is_later is called, such as in sync_file_merge. In other drivers, fence creation and cmdbuf submission normally happen atomically, mutex_lock(); fence = dma_fence_create(...,

This smells like https://github.com/gsliepen/tinc/pull/104 - are you sure you're using latest HEAD? How often does this occur? Does it correlate with other events such as nodes joining or leaving? Does it occur more often if you reduce the value of KeyExpire? On 22 December 2015 at 16:10, Florent B <florent at coppint.com> wrote: > Hi, > > With latest Tinc 1.1 git code, I have

If the fence context has notifies enabled, each of the fences' FENCE_FLAG_SIGNALED_BIT will be updated from the interrupt. We can rely on this status for reporting the current fence_is_signaled() and so avoid an expensive uncached read. Signed-off-by: Chris Wilson <chris at chris-wilson.co.uk> Cc: Ben Skeggs <bskeggs at redhat.com> Cc: dri-devel at lists.freedesktop.org Cc:

I've being using the same non-nat config for a 7960 for about a year with no issues. I have just upgraded cvs head from the 16th of July to todays. About a minute after I make a call I get the message chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission 000b46a0-8661007f-5c457e72-2b0dba2c@192.168.123.20 for seqno 102 (Critical Response) I don't get this for other

I've seen other request info on this config before. I've got the system up and running, Zap/Modem has been working for several days, no errors/warnings. Simply added info for broadvoice and I can dial into my system via the remote, but dial out from my box renders the following: I get one ring on the remote line, then on the console chan_sip.c:675 retans_pkt: Maximum retries exceeded

+0100 From: Daniel Schall <tinc-devel at mon-clan.de> Date: Thu, 6 Jan 2011 17:00:35 +0100 Subject: [PATCH] Improved PMTU discovery diff --git a/lib/dropin.c b/lib/dropin.c index 52fb5b8..2b803b1 100644 --- a/lib/dropin.c +++ b/lib/dropin.c @@ -165,8 +165,8 @@ #endif =20 #ifdef HAVE_MINGW -int usleep(long usec) { - Sleep(usec / 1000); - return 0; -} +//int usleep(long usec) { +//

Hi, I am a big fan of your project. Recently, I was trying to upgrade from tinc 1.1pre9 to pre10, and activating the experimental protocol, but a few things are not working. First of all, required libraries are missing in the windows installer, e.g. zlib.dll. Luckily, I still got them lying around from earlier versions (I don't remember, where I got them exactly.) I don't recall

CentOS Errata and Security Advisory YYYY:NNN <url> The following updated files have been uploaded and are currently syncing to the mirrors: files: ----------------------------------------------------------- The above will present the format of CentOS-4.2/alpha CESA-announcements. Here the 'alpha' is for axp and arch is labeled as 'axp' on subjects as 'alpha'

I'm running tinc 1.1pre11 with AutoConnect set to 'yes' and I recently started seeing lots of these messages on my VPN and cannot connect to various hosts from other hosts: (I have obscured the hostnames and vpn name, but otherwise this is a direct paste from syslog) Aug 19 14:51:51 AAA tinc.nnn[2217]: Got REQ_KEY from XXX while we already started a SPTPS session! Aug 19 14:51:54 AAA

On Mon, Feb 22, 2016 at 11:46:28AM +0100, Florent B wrote: > I still have this "bug"... any idea of what's going wrong ? There are some situations where keys are being renegotiated that can cause two peers to be very briefly out of sync. If that happens when there is VPN traffic, it could cause that message to pop up. If it only happens once in a while and you don't notice

Hello, Guus Sliepen, on Wed 02 Dec 2015 13:53:37 +0100, wrote: > That's great! It would be good though to split > handle_incoming_vpn_data() into a function that does the actual > recvfrom/mmsg() and one that processes each individual packet, to reduce > the level of indentation and make the functions a bit more readable. > Then I'll merge it :) Here it is. Samuel

10.12.2015 03:35, Samuel Thibault wrote: [] I suggest reducing ifdeffery in handle_incoming_vpn_data(), especially the error checking code. The function isn't that large now, it might be much better to have two different implementations. Like this (untested, patch attached): void handle_incoming_vpn_data(int sock) { #ifdef HAVE_RECVMMSG #define MAX_MSG 256 vpn_packet_t

We're trying to get a SIP peer going between our asterisk box and our provider. It should then ring our phone. The call does come in and it does execute the extension in the dial plan. But the provider says they never get a 200 OK back and therefore they send another INVITE and then after a few seconds drop the call. Here's our setup: sip.conf [ngt-trunk] type=peer qualify=yes port=5060

I have the following scenario: PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

Hi Jayson, I forward your request to the right place. Carlos (or some other) will have an answer, or at least some questions for you. -- Arnaud 2008/6/27 Jayson Anderson <jayson.anderson at gmail.com>: > Greetings Arnaud, > > I wrote you some time ago about whether NUT would support the Belkin > F6H550-UPS device in the future. > > You had replied and mentioned to send

We have an interesting issue: One of our providers has two softswitches. Calls coming from the first one are handled fine by asterisk, calls coming from the second one and going through the first one are euhm... dropped half a second into the RTP stream. I have opened a ticket at Digium for it: http://bugs.digium.com/view.php?id=10449 The output of "sip debug" is funny from line

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I have a remote installation that connects via IAX from my office pbx. When I call an extension on the remote pbx, after the dial period, the call is terminated. Nothing I do in configuration of that extension seems to matter: -- Executing NoOp("IAX2/netconcepts@nnn.nnn.nnn.nnn:4569-5", ""Dial 710"") in new stack -- Executing