similar to: compile error (still having problems)

I'm trying to compile the last * CVS version and I got this error: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-09/10/03-18:47:18\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Firstly, amazing software, props to all the developers. I'm trying to compile the latest asterisk cvs checkout and keep getting an error which I can't solve, any help would be much appreciated - make[1]: Leaving directory `/usr/src/asterisk/stdtime' if [ -d CVS ] && ! [ -f .version ]; then echo CVS-HEAD-05/17/04-16:45:34 > .version; fi for x in res channels pbx apps

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o

I am instaling asterisk on Fedora core 3. I have instaled zaptel-1.2.3, libpri-1.2.2, but when I am instaling (make install) asterisk I have the following error: .................... _GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_zapscan.o app_zapscan.c gcc -shared -Xlinker -x -o app_zapscan.so app_zapscan.o gcc -pipe -Wall

hello i have a small problem in installation of asterisk can any one tell me what is the solution gcc -shared -Xlinker -x -o app_zapscan.so app_zapscan.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-HEAD-05/26/05-20:43:39\"

Hi! I just installed Asterisk 0.4.0 with all the default options, and the configuration samples it has. When I try to dial from an h323 client (gnomemeeting) I get this message on the messages file: Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream): File demo-congrats does not exist in any format Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile):

Hi All I just upgraded to v1.0.1 so that I could try out the new SMS feature. When I did this, I noticed that my outbound SIP call to Broadvoice was very choppy, so I hung up. The system would not release the call, so I restarted the server. When it came up, it started loading * then said threw up a segmentation fault in res_crypto.so. It kept respawning until it ran out of RAM. I restarted

hello can u tell me what is the problem in my asterisk or linux why i am getting this error while make. PTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-HEAD-05/26/05-20:43:39\" -DASTERISK _VERSION_NUM=999999 -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIB DIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\"

am compiling h323 support using channels/h323/ error am getting is this any pointers chan_h323.h:30: warning: `struct sockaddr_in bindaddr' defined but not used g++ -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L/root/pwlib/lib -lpt_linux_x86_r -L/root/openh323/lib -lh323_linux_x86_r -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat /usr/i486-suse-linux/bin/ld: cannot find

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [asterisk@jonux h323]# make clean rm -f *.o *.so core.* [root@jonux h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN

Hello All. 1. I'm at first install ASTERISK (debian sid), and try to to make call by Kphone 3.1. For all my attempt I find in logs: Apr 18 16:05:17 NOTICE[81926]: File pbx.c, Line 999 (pbx_extension_helper): Cannot find extension context 'default' I check in extention.conf [default] section exist and has pattern which I send (for example 's', '1234'). In CLI I try to

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hello, I have VPC (Voice Pulse Connect) and NuFone for providers and I have setup modules.conf with the registered (Digium) G.729 Codec such as: load => codec_g729a.so load => res_crypto.so With both sip/iax2 configuration disallow=all is first and then allow=g729 is next (allow=ulaw,allow=alaw,allow=gsm are next after allow=g729) and it always dials via ulaw. Why is this happening? Josh

Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both the software but when i start Asterisk, it exits with the following dump. Error Text Start========================= [res_crypto.so] => (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key

I'm trying to enable res_crypto on a 1.4 installation, but menuconfig says ssl is needed. I've installed openssl, openssl-devel, openssl-perl but it's still not happy. Anyone know what else is needed? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090916/d21c2aca/attachment.htm

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

Hi All, I am a newbie and having some trouble getting Asterisk to work. I have checked out zapata zaptel libpri and asterisk (both v1-0_stable and regular--in separate directories). All built according to the documentation I have found and installed correctly. I have modified/created zaptel.conf zapata.conf as attached. I get the following messages/error during asterisk startup. ==

I'm running: OS: FreeBSD 6.0 Asterisk: 1.2.4 Installing OH323: 0.7.3 I have this error when compiling chan_oh323.c: In function `reload_config': chan_oh323.c:4677: warning: implicit declaration of function `sscanf' chan_oh323.c: At top level: chan_oh323.c:3244: warning: 'update_call_ids' defined but not used gcc -shared -Xlinker -x -g -o chan_oh323.so chan_oh323.o