similar to: Looking for a configuration guru to collaborate with

Here's a segment of my dialplan, I'm working on the freenum/ISN functionality: same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) same => n,GotoIf($["${isnresult}" != ""]?:fn-CONGESTION,1) ; set up our outgoing call state same => n,Set(SIPFROMUSER=${CALLERID(num)}) same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" ==

I've recently decided to spend idle cycles while waiting for various Astlinux platform builds to complete on making the contents of asterisk/config a little more complete, a little more useful, a little more real-world... I started looking at the possibility of taking JTodd's ISN Freenum cookbook example and updating it: http://freenum.org/cookbook/#ASTERISK_CONFIG but ran into some

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.

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

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,

On Fri, 8 Dec 2023 at 07:39, Philip Prindeville <philipp_subx at redfish-solutions.com> wrote: [...] > 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 non-shell accounts? sftp should work regardless of the user's shell since it is invoked as a ssh subsystem

The freenum.org beta continues to roll forward. If you have an Asterisk or SER SIP gateway/proxy, please see if you can make some sense of the examples below and install them in your system. Your users will hopefully be able to dial toll free numbers in various nations just like they dial regular numbers in those same country codes. I'd like to get some additional people trying to make

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

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

The freenum.org project wants to use your trunks! The freenum.org project is an ENUM parallel tree, which has as an eventual goal the distribution of ENUM numbering in nations or areas which due to political or other issues are not able to get secure, inexpensive, or functional ENUM capabilities. As a preliminary round, we're putting toll-free gateways into the system for various nations.

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

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

Hi. I have a Supermicro 5018D-FN4T (Xeon D-1541 based SBC) that I use for virtualization. I?m running Centos 7.3 on it (updated), with the CentOS-QEMU-EV.repo repository as the source for virtualization packages. I run an Ubuntu 16.04-2 guest VM on it, which is ordinary enough. What?s perhaps less ordinary is that I?ve attached a Lexar Media, Inc. ?Lexar Professional Workflow CR1 CFast 2.0 USB

> On Apr 10, 2023, at 7:24 AM, Darren Tucker <dtucker at dtucker.net> wrote: > > On Mon, 10 Apr 2023 at 07:07, Peter Stuge <peter at stuge.se> wrote: >> >> Brian Candler wrote: >>>> What's odd is that the length is *always* 1231976033 (which is >>>> 0x496E7661 or "Inva" in ASCII). > > One thing that can cause this is

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:

Hello! > Hello! > > We have the server with RHEL 5.5 (64-bit) and need to connect many parallel users over ssh (OpenSSH). > Usually we use openssh-4.4p1, builded from the sources with changed "servconf.h" file by this type: > #define MAX_ALLOW_USERS 10000 /* Max # users on allow list. */ > #define MAX_DENY_USERS

On Fri, 26 Jun 2015, Ludovic Gasc wrote: > 1. What's the "official" notation of each line: "=>" or "=" ? In the > wiki of Asterisk, I see very often "=>", however, what's the reason for > both syntaxes authorized ? Historical ? I'm not 'official,' but I have a strong preference for just '=.' Using

I'm trying to build an Asterisk rpm from the supplied asterisk.spec file. Made numerous changes to get it to work. The architecture of the system I am building on is x86_64. I'd like to build for i686 though. I added a --target i686 to the rpmbuild line in the Makefile, but it looks like it's still requiring 64bit system libraries. When I try to install the rpm on the i686 machine, it