similar to: Zap channel calling back after hangup (due to polarity CID detection)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: --

Hi all, I have an existing panasonic analog pbx in use and a asterisk server with digium tdm400p(2 fxs and 2 fxo). channel 1 -> fxs -> telephone channel 2 -> fxs -> telephone channel 3 -> fxo -> extension 15 at panasonic pbx channel 4 -> fxo -> phone line from telco We call in to fxo (channel 4) and enter the ivr which prompt us to enter the extension number. After

> On May 31, 2017 at 6:10 PM Ralf Hildebrandt <Ralf.Hildebrandt at charite.de> wrote: > > > * Ralf Hildebrandt <Ralf.Hildebrandt at charite.de>: > > > So I added > > ssl_ca_file = /etc/ssl/certs/ca-certificates.crt > > > > But alas: > > May 31 16:50:24 mproxy dovecot: config: Warning: Obsolete setting in

After upgrading from 2.2.28-1~auto+45 to 2.2.29-1~auto+25 I'm gettings this: May 31 16:44:31 mproxy dovecot: auth: Fatal: passdb imap: Cannot verify certificate without ssl_ca_dir or ssl_ca_file setting May 31 16:44:31 mproxy dovecot: master: Error: service(auth): command startup failed, throttling for 8 secs May 31 16:44:31 mproxy dovecot: imap-login: Disconnected: Auth process broken

Hi, this is a follow-up on my email from august (http://lists.cs.uiuc.edu/pipermail/llvmdev/2011-August/042737.html). i have, finally, released my OpenCL backend and control-flow restructuring framework for LLVM (AST-Extractor, or short axtor). The framework restructures function CFGs such that they can be expressed entirely without GOTOs or switch/loop-trickery. Hence, making it possible to

hello,sir, i need your help,i have cloned the axtor code from the bitbucked (the url:https://bitbucket.org/gnarf/axtor.git),but i can not compile it.so can you help me to compile it?my E-mail:tshping at 163.com,so if you give me some advice,send me e-mail to there.thx so much. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Micah, i just had a quick look at your structurizer. Here is what if found (correct me, if i am mistaken): * Our approaches for handling Loops with multiple exits are identical. ("Loop-Exit Enumeration") * Axtor implements Controlled-Node Splitting and can cope with irreducible control-flow. (http://cardit.et.tudelft.nl/MOVE/papers/cc96.ps) * Axtor translates switches to cascading

hi: In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i added polarity reversal property but for fxo number 4 i didnt add polarity reversal property but it still giving me on cosole that fxo number 4 is polarized (because the line on fxo number 4 is not polarized). what i want to do is to not let polarity reversal take effect on fxo number 4. that what i have in my

Hi all, I have a TC35 and am keen to see if anyone has both voice and sms working from Asterisk through this device? Google tells me that a few people have theorised about it, I can't find anyone claiming to be doing it. What would be the best way to put it into practice? Build a new channel for it? Thanks Andrew

Simon, Have you looked at the control flow structizer that we have in the Open Source AMDIL backend? > -----Original Message----- > From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] > On Behalf Of Simon Moll > Sent: Monday, March 05, 2012 1:01 PM > To: llvmdev at cs.uiuc.edu > Subject: [LLVMdev] OpenCL backend for LLVM > > Hi, > > this

Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1

Hi! I want to get audio from the PSTN before the call is answered so I don't miss when the called phone is busy or if there is some error (like the phone is unavailable or is wrong, etc) and hear the ringing from my telco. I have polarity reversal in my telco for incoming and outgoing calls. If I set answeronpolarityswitch=yes then I get no audio until the call is answered. If I set it to

Hello, This is a little off-topic. I have an Ericsson FCT f251m, according to the specs it supports call signalling through polarity reversals and loop break, but it's currently disabled. On my PSTN line, my TelCo does send polarity switchs to signal answer and hangup (answeronpolarityswitch=yes and hanguponpolarityswitch=yes in zapata.conf), and asterisk detects it alright. I've been

The person that wrote our structurizer agrees with your analysis. Too bad the licenses are incompatible, it would be nice to merge similar efforts. > -----Original Message----- > From: Simon Moll [mailto:simon.m.moll at googlemail.com] > Sent: Tuesday, March 06, 2012 2:49 AM > To: Villmow, Micah > Cc: llvmdev at cs.uiuc.edu > Subject: RE: [LLVMdev] OpenCL backend for LLVM >

Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]:

Hi, I have a problem with polarity reverse on answer I use asterisk 1.4.30 linux kernel version 2.6.27 dahdi version 2.2.1 and analog card is Sangoma a400 with fxo ports this is my config [trunkgroups]

Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

Hi, I am writing a decompiler. I was wondering if some of LLVM could be used for a decompiler. There are several stages in the decompiler process. 1) Take binary and create a higher level representation of it. Like RTL. 2) The output is then broken into blocks or nodes, each block ends in a CALL, JMP, RET, or 2-way or multiway conditional JMP. 3) The blocks or nodes are then analyzed for