similar to: TDM400P hangup / ringing detection problem

-----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 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 Is it possible to enable/disable the zaptel echo-canceller from the dialplan? The reason I ask is that I want the echocanceller active on all calls except when someone is sending a fax. The simplest way would be to disable it on incoming calls to the fax numbers and leaving it on on all other calls. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp

-----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 there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a "normal" phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very

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

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

I use both ports on my cisco ata-186. I run them using ulaw. Today I made numerous calls using my analog phone on port 2. I picked it up about an hour after the last call I made and the line was dead. There is no power at all over the phoneline to the phone, and the red light doesnt light up. The configuration is verified as unchanged. Has anyone seen this problem before. I was

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, I'm relatively new to the subject - so pleace don't punish me for idiotic questions. ;-) Can Asterisk act like a normal Sip phone and e.g. connect to another sip-gateway? Background: There is a new german company at: http://www.sipgate.de (sorry German only page) They offer a a gateway between a real telephone number and

https://bugs.freedesktop.org/show_bug.cgi?id=38673 Summary: all object is black Product: Mesa Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: major Priority: medium Component: Drivers/DRI/nouveau AssignedTo: nouveau at lists.freedesktop.org

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

It said 'include zapata-channels.conf', where this line wasn't commented bij the ';'... Could you post me a working example of such a config (or a part of it, for the X100P cards...? Thanks guys! Message: 9 Date: Sat, 19 Mar 2005 18:04:26 -0500 From: "Jeff Glassman" <jrglass@columbus.rr.com> Subject: [Asterisk-Users] newbie question To:

Hi, I'm calling asterisk with a swedish PSTN-phone line with caller presentation (DTMF) activated. I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay with this release. I use a TDM800P 8 channel PSTN card working as answering phones (I connect a phoneline with carrier signal to my TDM-card). Using zaptel-1.4.12.1. I verified that the DTMF tones of the number

* 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