similar to: make problem.

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

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

I have some problems when trying to install Asterisk on Mandrake. gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs gthread` /usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackware-linux/bin/ld: cannot find -lXext collect2: ld returned 1 exit status make[1]: *** [pbx_gtkconsole.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/pbx' make:

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,

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 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

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

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

This patch is against libguestfs's old copy of p2v, but the same patch ought to also apply against virt-p2v. Rich.

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

I got the error below while compiling asterisk. Please offer me some help. hubert for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x depend || exit 1 ; done make[1]: Entering directory `/etc/asterisk/asterisk-0.7.2/res' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/etc/asterisk/asterisk-0.7.2/res' make[1]: Entering

i'm having this error message when trying to run the agents-feature Sep 2 17:37:40 WARNING[10445]: pbx.c:1645 pbx_extension_helper: No application 'AgentsLogin' for extension (from-internal, 28, 1) while chan_agent.so is beeing loaded i still don't seem to have access to the commands like agentlogin or agentcallbacklogin. my agents.conf and queues.conf are configured correctly

Hi Guys, I've attempted to get this moh-native thing to work with no success. I've reviewed wiki, mantis and e-mail postings and I'm confused. The latest I've read is native moh should be in asterisk-addons in format_mp3, but what version will it work with? I've tried asterisk 1.0.1, 1.0.5, addons 1.0.1, 1.0.4 and also -r stable CVS. I followed the wiki example with

Hi *, I have set up a very simple asterisk configuration where I intend to be redirected to the voicemail whenever I dial 100 with my kphone SIP client. The problem is that asterisk can not find the stream 'vm-theperson'. I have made a non-standard installation (since I am just testing), and that file is located in /mnt/tr2/fake_root/installed/usr/local/var/lib/asterisk/sounds. 1. How

Compiled Asterisk from FreeBSD port (0.9.0_2) When I start asterisk it uses 100% cpu. Searches on Google say to comment the noload => chan_oss.so in modules.conf But this is already commented. Make.conf contains some optimizations. modules.conf: ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes ; ; If you want, load the GTK console right away. ;

Currently we have problems with our asterisk server connect with an Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 on RedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL. The Diva server is running in version 7.5 Can someone help us in reolving the errors with incoming calls? When we try to call to an extension we get these messages in the CLI: -- creating pipe for

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