similar to: Enabling/disabling zaptel echo-can from dialplan.

-----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: --

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi!. I am having problems with getting asterisk to detect when someone hangs up. I have a TDM400P with one FXO module connected to my telco, and also a FXS-module connected to my phone. The FXS-module detects hangups just fine, but I can't get the FXO to detect them. I am pretty sure i have disconnect supervision on my phoneline since when I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I live in Sweden and I am having problems getting asterisk to properly detect when a caller hangs up. And yes, I DO have disconnect-supervision on my line. Also asterisk sometimes misinterprets the disconnect-signal as another incoming call. This usually happens if I hang up first and then when the caller hangs up, asterisk treats it as a new

-----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

Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine. I bought the card mainly to get caller ID to work properly in Sweden, and that works just fine. However, if the called or calling party hangs up after I hangup my SIP channel, polarity CID detection kicks in and dials a couple of signals to my incoming context. This happens with Asterisk 1.0.6 and CVS-HEAD. I have tried

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:

> 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 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

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

* 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 /etc/dovecot/conf.d/10-ssl.conf:36: ssl_ca_file has been replaced by ssl_ca = <file > > Gnarf! As you can see I do HAVE ssl_ca in my doveconf -n output! > >

* Aki Tuomi <aki.tuomi at dovecot.fi>: > > > 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 /etc/dovecot/conf.d/10-ssl.conf:36: ssl_ca_file has been replaced by ssl_ca = <file > > > > > > Gnarf! As you can

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

> On June 1, 2017 at 1:42 PM Ralf Hildebrandt <Ralf.Hildebrandt at charite.de> wrote: > > > * Aki Tuomi <aki.tuomi at dovecot.fi>: > > > > > 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

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 >

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, 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

Hi, I am trying to validate a setup with a sangoma A116 card (16PRI in one card). I currently have two machines set up, each with a sangoma A116 card. Those are interconnected with crossed PRI cables. One of them is in NT mode, the other in TE. The ports are configured in E1 mode, asterisk is all set up and I can make calls between them, filling up all 480 (16 * 30) channels. Now, I learned

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

Dear, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? Thanks a