search for: pbx_wilcalu

This is cvs version from yesterday. There is small problem with that version. When starting I've got: [...] [skipping pbx_gtkconsole.so] [pbx_spool.so] => (Outgoing Spool Support) /var/spool/asterisk/outgoing [pbx_wilcalu.so] WARNING[1024]: File loader.c, Line 226 (ast_load_resource): /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_select WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module pbx_wilcalu.so failed! Anyone had the same problem? I'm sure that I've changed all of...

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: ast_pthread_create Jul 26 22:59:18 WARNING[31473] loader.c: Loading module pbx_wilcalu.so failed! If I "noload" the mo...

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 enterprise zap...

start it with asterisk -vvvgc bkw ----- Original Message ----- From: "Arjan" <arjan@inventionz.org> To: <asterisk-users@lists.digium.com> Sent: Sunday, July 11, 2004 12:27 PM Subject: [Asterisk-Users] Asterisk on FreeBSD 4.10 dies > Hi All, > > I'm pretty green to Asterisk. I'm trying to work towards a basic setup > with a couple of Cisco 7960's

...run\" -DASTSPO OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT -fPIC -c -o pbx_wilcalu.o pbx_wilcalu.c gcc -shared -Xlinker -x -o pbx_wilcalu.so pbx_wilcalu.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-07/22/03-22:29:39\&...

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 [mailto:asterisk-users-bounces@lists.digium.com] On Behalf O...

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '2d57437068f8800f671168c74e01eae1@210.9.49.249' of Reque...

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

...an I have somebody call in, then have the system call a list of numbers (i.e. the other teams) and play to them what the caller is saying? If that isn't possible, can I have the caller record a sound, then play that to the other teams all at once (or one at a time)? I see there are modules pbx_wilcalu and pbx_spool that look like they might do something of the sort, but I can't find any docs on this--I can create a new module if there isn't any functionality to do something like this, as I'm a programmer, too, but I'm trying to avoid duplicating work already done... Thanks f...

...I normaly installed asterisk, created an * user, moved the binaries to /usr/bin and chowned all the files and directories mentiont in the * manual (handbook-draft.pdf) Now I can start * but I get the following warning (which I don't get if I run it as a root): Feb 14 19:10:53 WARNING[213006]: pbx_wilcalu.c:69 autodial: Autodial: Unable to open file ~ == Parsing '/etc/asterisk/enum.conf': Found I don't know if * really works - I have't tired jet - can anybody tell me which file * want's to access? ( I looked in the source but I'm not that familiar with the code) Thank'...

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. ;

...sm.so load => format_h263.so load => format_ilbc.so load => format_jpeg.so load => format_pcm.so load => format_pcm_alaw.so load => format_vox.so load => format_wav.so load => format_wav_gsm.so load => cdr_csv.so load => pbx_config.so load => pbx_spool.so load => pbx_wilcalu.so load => res_adsi.so load => res_crypto.so load => res_indications.so load => res_monitor.so load => res_musiconhold.so load => res_parking.so load => app_adsiprog.so load => app_agi.so noload => app_alarmreceiver.so load => app_authenticate.so load => app_cdr.so...

...n't save CDR for 0 app_image.so Image Transmission Application 0 format_mp3.so MP3 format [Any rate but 8000hz mono opt 0 app_festival.so Simple Festival Interface 0 app_privacy.so Require phone number to be entered, if n 0 pbx_wilcalu.so Wil Cal U (Auto Dialer) 0 app_substring.so (Deprecated) Save substring digits in a 0 chan_skinny.so Skinny Client Control Protocol (Skinny) 0 format_sln.so Raw Signed Linear Audio support (SLN) 0 app_zapateller.so Block Tel...

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

...for /usr/lib/libpri.so.1 Reading symbols from /usr/lib/libtonezone.so.1...done. Loaded symbols for /usr/lib/libtonezone.so.1 Reading symbols from /usr/lib/asterisk/modules/pbx_config.so...done. Loaded symbols for /usr/lib/asterisk/modules/pbx_config.so Reading symbols from /usr/lib/asterisk/modules/pbx_wilcalu.so...done. Loaded symbols for /usr/lib/asterisk/modules/pbx_wilcalu.so Reading symbols from /usr/lib/asterisk/modules/pbx_spool.so...done. Loaded symbols for /usr/lib/asterisk/modules/pbx_spool.so Reading symbols from /usr/lib/asterisk/modules/app_dial.so...done. Loaded symbols for /usr/lib/asteris...

...0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_playback.so Trivial Playback Application 0 app_dial.so Dialing Application 0 pbx_spool.so Outgoing Spool Support 1 pbx_wilcalu.so Wil Cal U (Auto Dialer) 0 pbx_config.so Text Extension Configuration 0 chan_zap.so Zapata Telephony w/PRI 0 chan_phone.so Linux Telephony API Support 0 chan_skinny.so Skinn...

...No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 23/09/2005 ------------------------------ Message: 25 Date: Tue, 27 Sep 2005 16:36:21 +0900 From: Pikoro <webmaster@psphacks.net> Subject: [Asterisk-Users] pbx_wilcalu.so: undefined symbol: To: asterisk-users@lists.digium.com Message-ID: <4338F675.9030702@psphacks.net> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Anyone run into this? This is from the latest 1.2.0 beta1 tarball. Got it all compiled, but this undefined symbol is stoppi...

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

...n_oss.so] => (OSS Console Channel Driver) WARNING[16384]: File chan_oss.c, Line 429 (soundcard_init): Unable to open /dev/dsp: No such device [chan_phone.so] => (Linux Telephony API Support) [chan_zap.so] => (Zapata Telephony w/PRI) [pbx_config.so] => (Text Extension Configuration) [pbx_wilcalu.so] => (Wil Cal U (Auto Dialer)) [pbx_spool.so] => (Outgoing Spool Support) /var/spool/asterisk/outgoing [app_dial.so] => (Dialing Application) [app_playback.so] => (Trivial Playback Application) [app_voicemail.so] => (Comedian Mail (Voicemail System)) [app_directory.so] => (E...