similar to: [Bug 1964] New: QoS/DSCP names false translated to ToS hex value

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=1733 Summary: Enhance support for QoS (ToS) by supporting DSCP/CS and adding option Product: Portable OpenSSH Version: 5.4p1 Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P2 Component: ssh AssignedTo:

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,

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

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.

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:

Trying to figure out how to prefer voip traffic on a dsl line. Found a great howto: http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk but I'm trying to figure out the relationship between the tos of iax.conf and tos of tc from Iproute2. my traffic goes from my linux router to a CPE cisco box. I understand Cisco uses tos ( usually referred to as DSCP, just to keep us on

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

I am assuming that the TOS values directly map to DSCP values in the ip header. Is this a correct assumption? If so, can someone tell me the correct setting to set call control packets with a DSCP of AF31(011010) and media with EF(101110)? So would the setting for AF be TOS=46?? Is it possible to mark the media and call control separately?? -------------- next part -------------- An HTML

https://bugzilla.mindrot.org/show_bug.cgi?id=3634 Bug ID: 3634 Summary: IPQoS default should be changed to "none" Product: Portable OpenSSH Version: 9.5p1 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: Miscellaneous Assignee:

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

On Wed, 3 Apr 2019, Adam Eijdenberg wrote: > On Wed, Apr 3, 2019 at 9:43 AM Damien Miller <djm at mindrot.org> wrote: > > Could you try turning off IPQoS setting (ssh -oIPQoS=none ...) and seeing > > whether that makes any difference? > > Yes - setting -oIPQoS=none on the client allows for a successful > connection to the server. > > Would you still like me

Added the line as above: 'IPQoS lowdelay throughput', restarted the SSH service, tried SSH-ing and Git-ing... same error as before: 'packet_write_wait: Connection to X.X.X.X port 22: Broken pipe'. ________________________________ From: Zach Cheung <kuroro.zhang at gmail.com> Sent: 29 August 2018 07:17 To: ohadfjunkbox at outlook.com Cc: dtucker at dtucker.net;

I''m successfully using HTB + GRED to shape traffic based on the DSCP field. I would like to strip the DSCP and possibly replace it with normal ToS bits on egress traffic leaving my network. Leaving DSCP set is pointless, and could potentially cause problems with some ISPs that use DSCP internally I suppose. Setting ToS bits would seem ideal as most networks still honor it to varying

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

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

On 07/12/23, Philip Prindeville (philipp_subx at redfish-solutions.com) wrote: > 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. Off topic, and assuming the .p12 bundles need to be post-processed by clients for use by ssh,