similar to: "friendly" dial tone frequency combinations

Hi all, Can anybody explain what these values exactly mean. As you all know its the dialtone value on an SPA3000 of linksys. 350@-19,440@-19;10(*/0/1+2). Can anybody help me how to write this code for a dialtone of frequency 425 which is continous. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, We have an application running over samba share that creates a lot of 0-length files, and eventually throws an error because one of more of the files could not be created or does not appear to be created. At the same time in the samba log there are many messages like [2015/02/02 14:02:05.545710, 0, pid=32169, class=locking]

For anyone that is interested in looking at HIP, I have posted the latest HIP build on my server: http://htt-consult.com/~rgm/hip/ You don't need the src rpm, the others will do. Since we do not have the IPsec ESP BEET mode patch (http://www.ietf.org/internet-drafts/draft-nikander-esp-beet-mode-09.txt) for the 2.6.18 kernel (it is in the 2.6.27 kernel), we have to run HIP in userspace.

Hi people! Will this filter work? # the traffic for MAN - match IP & MARK $tc filter add dev $EXT1 parent 1: protocol ip prio 0 handle 0x1 fw u32 match ip src $IP flowid 1:B$hIP $tc filter add dev $INT1 parent 1: protocol ip prio 0 handle 0x1 fw u32 match ip dst $IP flowid 1:B$hIP # the traffic for Internet - match IP only (what is left after MAN filter) $tc filter add dev $EXT1 parent 1:

Does anyone know if it is possible to use the following disconnect tone setting with an x100p card? Disconnect Tone: 350@-19,440@-19;4(.25/.25/1+2) This tone was written for a Sipura SPA-3000 for a Panasonic KX-TD1232. The Panasonic does not support disconnect supervision, so this tone is the only thing that will detect a disconnect. It is not a standard fast busy or offhook tone.

Hi, I have 2 Asterisk systems and a unique scenario where I need to play a particular tone on Asterisk1 and identify the same tone on Asterisk2. Following is my call flow, Asterisk1(Plays audiofile1,Wait for 2 Sec,Plays a tone,Plays audiofile2) -> PSTN -> 3rd Party CONFERENCE SYSTEM <- PSTN <- Asterisk2(Record audiofile1,Wait for a tone,Record audiofile2). A few points to keep in

hi,? I'm working on a research project where we're looking at the changing resonances in femur bones during hip replacement operations. basically, I've got a spreadsheet with one column listing frequencies in 5Hz bands and 3 columns showing the amplitude of that frequency when the bone is loose, medium to tight (referring to the tension of the reamer/chisel creating a cavity for the

--On Thursday, February 03, 2005 2:20 PM -0500 Andrew Thompson <asteriskuser@aktzero.com> wrote: > What you are seeing with these bargain providers is they have a clause in > their contract that says they own the number, not you. It is a lock, and > it ought to be illegal, but sadly, it's probably not. If you choose one > of these companies that doesn't allow you to

LLVM Weekly - #222, Apr 2nd 2018 ================================ If you prefer, you can read a HTML version of this email at <http://llvmweekly.org/issue/222>. Welcome to the two hundred and twenty-second issue of LLVM Weekly, a weekly newsletter (published every Monday) covering developments in LLVM, Clang, and related projects. LLVM Weekly is brought to you by [Alex

Hi, I'm Andrew. (Hi Andrew) I'm a toll-free number junkie. I've had an account with iax.cc/sixtel for about a week, and every few days, I find myself sitting at the DID menu clicking the link that reads "Click here to get a random toll free number". I have three toll-free numbers now, and I don't know if it will stop... Is there any hope for me? -- Andrew

I do not know enough about cling, but I like what you describe very much, am particularly intrigued about how your approach could also be appropriated to do ahead-of-time constexpr metaprogramming as well, which also involves incrementally adding declarations to the translation unit. Dave > On Jul 9, 2020, at 11:43 PM, JF Bastien via cfe-dev <cfe-dev at lists.llvm.org> wrote: > >

This patch fixes a few non-ASCII characters which were displayed wrong, adds news about Pono and adds a few artists to the links page --- comparison.html | 10 +++++----- feeds/feed.xml | 18 +++++++++++++++++- links.html | 3 +++ 3 files changed, 25 insertions(+), 6 deletions(-) diff --git a/comparison.html b/comparison.html index 8e6b74b..daa0906 100644 --- a/comparison.html +++

iaxComm is a cross-platform IAX2 softphone available for Win32 and Linux. Win32 and Linux binaries as well as the LGPL source are available at: http://iaxclient.sourceforge.net Recent improvements are a less cluttered user interface, audible ringback and audible outgoing ring, and of course IAX2 protocol support. iaxComm is based upon the wxWindow GUI framework and compiles on Microsoft

My colleague, doing the HIPL development, principally uses ubuntu and although he is now building the Fedora/Centos rpms for me, he leaves much of the testing to me..... Anyway for any kernel prior to 2.6.27, I have to run the hip code in userspace, not kernel space. This is due to the ESP BEET patch. There is a specific hipd for userspace, but the hipfw requires a different option. For a

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

In my testing of HIP, there is reference to use 'nc6' for a simple TCP character echo server. No such animal in Centos, and after a bit of digging I find this refers to Netcat6 which seems to be a dead app? Is it available as an rpm somewhere for Centos? My searching is coming up empty. Or just about any service that I can start on one system and connect from another that would

We have a test server that runs a single PIII 500MHz(256MB RAM) under Slackware, and we can get 12 SIP -> Zap calls running on it just fine. Over that and we have seen intermittant errors like call quality and very high load spikes. You should be able to get at least 24 SIP -> Zap on that setup. Post on the list when you do max it out. It's always good to see capacity specs. MATT---

David Jones <dlj at google.com> writes: > >> * Provide C symbols as specified by the standards, but take advantage > >> and use C++ language facilities for the core implementation. > > Does this mean C programs would require a C++ runtime? If not, how will > the project ensure that? > > Shooting from the hip: no. Turning off exceptions, RTTI, and static

I know this has been covered more times than to mention and this is where I got most of my info from... But I am having issues with this. I can't seem to get the phone to register with *. This is being tested on a internal network right now. Here is the setup - sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context

Levente Kovacs writes: > I continuously get user authentication errors. I suspect that the problem is > with a password hashing. I used > > # doveadm pw -s SHA512-CRYPT > Enter new password: > Retype new password: > {SHA512-CRYPT}$6$1cCNWIMCTpuqmexO$cCq7nDEga1xza3967nZ.BFZaGWNoYwjnNfSCvZE/jGzYAY1hLWE1iHR1KYaO.mvVE4WVdDSrvPNtfwAUn/yk3/ Looks OK. > I have this at the