similar to: [Bug 1733] New: Enhance support for QoS (ToS) by supporting DSCP/CS and adding option

https://bugzilla.mindrot.org/show_bug.cgi?id=1964 Bug #: 1964 Summary: QoS/DSCP names false translated to ToS hex value Classification: Unclassified Product: Portable OpenSSH Version: 5.9p1 Platform: amd64 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh

https://bugzilla.mindrot.org/show_bug.cgi?id=1856 Summary: Wrong QoS naming and obsolete defaults Product: Portable OpenSSH Version: 5.8p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: Miscellaneous AssignedTo: unassigned-bugs at mindrot.org ReportedBy:

https://bugzilla.mindrot.org/show_bug.cgi?id=1963 --- Comment #5 from martin f. krafft <bugzilla.mindrot.org at pobox.madduck.net> --- With reference to http://bugs.debian.org/650512, which I just reopened, I am sorry to say that the bug persists in OpenSSH 6.0. -- You are receiving this mail because: You are watching the assignee of the bug. You are watching someone on the CC list of the

On Mon, 27 Aug 2018, Stuart Henderson wrote: > On 2018-08-27, Zach Cheung <kuroro.zhang at gmail.com> wrote: > > After upgrading my VMware Fusion (10.1.3) Arch Guest to the latest with > > OpenSSH upgraded from 7.7p1 to 7.8p1, found that ssh from macOS Sierra > > (10.12.6) host to Arch guest via local NAT port forwarding failed, but via > > Arch LAN IP worked,

http://bugzilla.netfilter.org/show_bug.cgi?id=795 Summary: RELATED doesn't accommodate multicast UDP solicitation resulting in unicast reply Product: netfilter/iptables Version: unspecified Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P5 Component:

I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from "astlinux" to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming

Here's the bug and proposed patch. It's pretty trivial. https://bugzilla.mindrot.org/show_bug.cgi?id=1856 Quoting RFC-2474: A replacement header field, called the DS field, is defined, which is intended to supersede the existing definitions of the IPv4 TOS octet [RFC791] and the IPv6 Traffic Class octet [IPv6]. [...] The structure of the DS field shown above is incompatible with

I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of

Anyone have a chance to test any of the various iPhone SIP apps? I see there are a few out there, but most of the iTunes reviews aren't sufficiently technical to be useful. Thanks.

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using

Hi, We have a CLI that certain users get dropped into when they log in. One of the things they can go is generate certificates (actually .p12 key/certificate bundles) that they will then scp out of the box from another host. Problem is that if their default shell isn't sh, ash, dash, bash, zsh, etc. then things break. Is there a workaround to allow scp/sftp to continue to work even for

Has anyone figured this out yet? Lots of places say to add the following to sip.conf of an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ;

For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk and my Sipura 942's, for instance... Not sure what these are... perhaps the qualify keepalives? In which case, I guess

Sorry for the crosspost (this was also posted to asterisk-at-uc-dot-org) but I haven't got a response. I have a cell phone added to a queue as a local extension (member => Local/299). I want the cell phone to be able to reject calls to the queue without the person sitting in the queue being hung up on, etc. The way my dialplan is set up, the person hits 1 to answer the call and any other

problem - occasional garbled calls, mostly remote users. T1 connected to PSTN, SIP over local LAN and internet to "remote users". NAT at local firewall and at remotes. There is no traffic shaping in place, no QoS. Most are Polycom phones, two Aastra's. Start with QoS on LAN switches? No 2x4's please, start with 1x4's. joe a.

https://bugzilla.mindrot.org/show_bug.cgi?id=1708 Summary: Bugs intended to be fixed in 5.4 Product: Portable OpenSSH Version: -current Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: Miscellaneous AssignedTo: unassigned-bugs at mindrot.org ReportedBy: djm

I was wondering under what conditions Asterisk will hand off a call to another switch. I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. Thanks, -Philip

Hello - What I want to do seems very simple - I want to make sure any H.323 traffic gets processed before anything else entering or leaving this network. The network has a videoconferencing device on the LAN at 192.168.16.4. A Linux firewall NATs an external IP Address to this internal address and I have appropriate SNAT and DNAT rules that work. The NAT and connection tracking rules all work

Sorry about taking so long to get back to you. The problem is sporadic and I've had other fires to put out first... Here's a PCAP of authentication failures: https://www.redfish-solutions.com/misc/kvm1.pcap > On Apr 9, 2023, at 1:21 AM, Brian Candler <b.candler at pobox.com> wrote: > > On 09/04/2023 02:20, Philip Prindeville wrote: >> What's odd is that the

On Tue, 25 Apr 2023 at 03:36, Philip Prindeville <philipp_subx at redfish-solutions.com> wrote: > > On Apr 10, 2023, at 7:24 AM, Darren Tucker <dtucker at dtucker.net> wrote: [...] > > Since you're using 9.1, the message could be an "Invalid free", since > > there was a double-free bug in that release :-( > > Forgot to ask: does this bug manifest