similar to: pbx_wilcalu.so undefined symbol

Displaying 20 results from an estimated 700 matches similar to: "pbx_wilcalu.so undefined symbol"

2004 May 09
2
[BUG] rsync 2.6.2
After upgrade from previous version I can't run rsync. 2004/05/09 10:40:54 [18630] rsyncd version 2.6.2 starting, listening on port 873 2004/05/09 10:40:54 [18630] rsync error: error in socket IO (code 10) at socket.c(466) strace shows that there is problem with listen() I've found patch on this lists: --- rsync-2.6.2/socket.c.orig 2004-05-08 23:25:11.979473336 +0200 +++
2003 May 16
0
asterisk, sip and SRV record
Can asterisk use SRV record from DNS? In example - I've got: _sip._tcp.gda.pl. SRV 0 0 5060 welbot.task.gda.pl. _sip._udp.gda.pl. SRV 1 0 5060 welbot.task.gda.pl. And it works with sipd (from Columbia) and vocal. I see that with asterisk it does not want to work... :-/ -- pozdr. Pawe? Go?aszewski --------------------------------- worth to see:
2003 Sep 04
0
oh323 <-> sip communication problem
I've got problem with connections h323 -> sip and sip -> h323. I've Cisco 7940 phone with sip soft and Netmeeting as h323 node. As gatekeeper I've gnugk and brand new asterisk from cvs + chan_oh323 0.5.5 When I call from Cisco (SIP) to h323 node by alias registered on gatekeeper and h323 node will answer the phone... I have on my Cisco still Ringing. Call termination, no
2001 Sep 08
1
force SSH1 and SSH2
This is small patch for scp. It allows to force SSH1 or SSH2. P.S.: give me Cc: - I'm not subscribed... -- --------------------------------- pozdr. Pawe? Go?aszewski --------------------------------- R.I.P. - rest in pieces ... -------------- next part -------------- --- ./scp.c.org Sat Sep 8 23:37:22 2001 +++ ./scp.c Sun Sep 9 00:07:36 2001 @@ -244,9 +244,11 @@
2004 Sep 21
3
FreeBSD 100% cpu
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. ;
2005 Jul 26
1
What does pbx-wilcalu.so do and why does it keep crashing my * box?
I downloaded the latest CVS a few days ago. It all compiled nicely on my new AAH platform. However, it won't start up. Investigation of my log files produces this; Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 WARNING[31473] loader.c: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol:
2003 May 21
6
chan_oh323.so: Segmentation Fault
Hi, I'm trying to get H323 support using asterisk 0.4.0 Unfortunately the pwlib and openh323 versions mentioned in the asterisk-oh323 readme file are no more available, and I had to use newer ones. Now I installed all libraries, but got a segemntion fault when starting asterisk while reading the chan_oh323.conf file. When I declare more than 9 gwprefix I get first a error "out of
2006 Jun 11
1
asterisk-1.2.9.1
hi ! i have installed asterisk-1.2.9.1 but am unable to run it i am getting this error "[pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed!" can anyone help me i have redhat linux
2003 Oct 22
2
Slackware 9.1 Install Help
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
2005 Jul 15
1
make problem.
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
2003 Jul 23
1
Newbie Help
Hi - after hearing others rave about * I thought I'd have a go - extract from a 'make' on a stock debian system as follows... (I tried to post the whole make up to this point but it was too big for the list) make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
2005 Jun 06
1
CLUELESS NEWBIE needs help making an outboundsip call to PSTN
Steve, 1) go to /etc/asterisk 2) open modules.conf for editing using vi 3) add this line: noload=pbx_wilcalu.so 4) Save the file 5) Restart asterisk Lightup the candles, open the Cabernet Savignon ( or whatever your prefernce) and call your girlfriend. ;) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com
2003 May 26
3
Cisco 7960 SIP speed dial
I'm having great success using Cisco 7960s with Asterisk but one Cisco 7960 specific function that's puzzling me is programming the 'line buttons' (6 buttons on the upper right side of the phone) for speed dial. I'm not using any Cisco applications like Call Manager and the phones are SIP. *All* of the documentation I've found over the last several days continues to refer
2003 Dec 22
2
Compile Problem
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
2004 Jan 09
1
Help with compiling
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:
2004 Jun 23
1
capi.so problem on startup
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
2004 Apr 26
3
Compiling asterisk
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
2005 Sep 02
1
No application 'AgentsLogin'
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
2005 Feb 13
1
MusicOnHold Native Mode, Please Clarify
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
2004 Jul 02
0
Problem locating stream files
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