similar to: Help with compiling

I'm looking for the patch that enables suppotr for the following lines in queue.conf: announce-holdtime = yes queue-youarenext = "queue-youarenext" queue-thereare = "queue-thereare" queue-callswaiting = "queue-callswaiting" queue-holdtime = "queue-holdtime" queue-minutes = "queue-minutes" queue-thankyou = "queue-thankyou" It

Hi, I can't compile Asterisk on a Debian machine. gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o

Hi, I'm trying to install Asterisk onto a fresh install of Slackware 9.1... I've installed all packages in A, AP, D, E, F, K, L, N... So basically what's needed for a text based system with development, networking, docs, libraries.. No X-Windows, no games, no TCL/TEX etc. Following the commands on the Asterisk website I checked out the CVS source and started compiling... Zaptel and

We have a strange make problem : In file included from /usr/include/gtk-1.2/gtk/gtkarg.h:31, from /usr/include/gtk-1.2/gtk/gtkobject.h:31, from /usr/include/gtk-1.2/gtk/gtkaccelgroup.h:35, from /usr/include/gtk-1.2/gtk/gtk.h:32, from pbx_gtkconsole.c:39: /usr/include/gtk-1.2/gtk/gtktypeutils.h:163: warning: function

I am trying to compile the asterisk and if fails at the end on: make[1]: Entering directory `/usr/src/asterisk-0.5.0/pbx' gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs gthread` /usr/lib/gcc-lib/i486-slackware-linux/3.3.2/../../../../i486-slackware-linux/bin/ld: cannot find -lXext collect2: ld returned 1 exit status make[1]: *** [pbx_gtkconsole.so] Error 1

Hi, I'm new to asterisk and try to get it work with capi.so. When I try to start asterisk with "asterisk -vvvvc" I get the following errors. I couldn't find any hint on the net what may be wrong in my configs. Has anybody got a hint? Here is the error output: [capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Jun 23

when i issue asterisk -vvv command i get this error please help regards Barbra [app_softhangup.so] => (Hangs up the requested channel) == Registered application 'SoftHangup' [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format 7 to 6, cost 50 == Registered translator 'lintolpc10' from format 6 to 7,

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

This is my second attempt trying to get help and I am hoping someone can. When the musiconhold extension is matched, Asterisk attempts to execute musiconhold and stops right away, this is what I gets: Executing MusicOnHold("OSS/dsp", "") in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Is there a file that

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

Hello! Asterisk "CVS-HEAD-08/06/04-14:55:13" won't start on two of three different Gentoo machines. This is the output of gdb: ultra asterisk # gdb /usr/sbin/asterisk GNU gdb 6.0 Copyright 2003 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions.

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Hello List! I finally got asterisk with capi working, and its already answering my call as well! :) Now i would like to call a number from my shoft phone (kphone). This is my extentions.conf: --- [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password

Hi Guys, We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL & Asterisk Addons. I have populated the "sip_buddies" table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Hello all! My basic problem is that we haven't been able to get realtime to use ODBC to store configuration data. Here are the details: We've installed Asterisk on a CentOS machine as follows: 1. Downloaded, compiled, and installed FreeTDS 0.63 2. Downloaded, compiled, and installed unixODBC 2.2.11 3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons, and Zaptel from CVS

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So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin