similar to: undefined symbol: cap_set_proc on several modules after installation from source

I have built Asterisk 1.4.4 on my Solaris 10 x86 box: LDFLAGS='-R/usr/sfw/lib -R/opt/csw/lib -L/opt/csw/lib -L/usr/sfw/lib' CPPFLAGS=-I/opt/csw/include ./configure -with-curl=/opt/csw --without-oss --without-vpb --prefix=/opt/asterisk-1.4 The build and install go fine but the asterisk executable reproducibly dumps core with a segmentation violation. If I start it as: asterisk -gc and

> Message: 1 > Date: Tue, 15 May 2007 23:01:24 -0400 > From: Frank Tarczynski <ftarz at mindspring.com> > Subject: [asterisk-users] Asterisk 1.4.4 reproducibly dumps core on > Solaris 10 > To: asterisk-users at lists.digium.com > Message-ID: <464A7404.5000706 at mindspring.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I have

Hi I have asterisk 1.8.18 with freepbx 2.10.1.9. I get an asterisk crash occasionally with the followingerror. It always seems to happen while paging. 16 spa508g phones 1 snom pa1 paging amp Kelly == Extension Changed 1101[ext-paging] new state Idle for Notify User 101 pbx*CLI> *** glibc detected *** /usr/sbin/asterisk: malloc(): smallbin

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

Hi, I've been having a little problem with my asterisk servers, I have 4 identical asterisk servers setup (same hardware, same OS, same config). Once in a while (once or twice a day) one of the server crashes on the cron job reload. But I realized this only happens on 3 of the 4 servers. Tried to spot the difference between that one server that wasn't crashing. The difference I found was

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

Dear ALL, Any clues or tips for the following gdb messages. [root@localhost asterisk]# uname -a Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct 29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux localhost*CLI> show version Asterisk CVS-HEAD-09/22/04-11:19:09 built by root@localhost on a i686 running Linux [root@localhost asterisk]# gdb asterisk core.13089 GNU gdb Red Hat Linux

I was wondering if it might be possible for an rsync developer to look over the attached patch (tested on Linux 2.4.24 against the rsync-2.6.0 release), and offer suggestions on how I could improve it. Basically I want to use Linux finer grained capabilities to retain only CAP_SYS_CHROOT & CAP_DAC_READ_SEARCH when rsync drops root privs. That way I can take whole system backups as a (mostly)

Hello using pjproject 2.5.5 using asterisk-certified-13.8-cert1 Compiled pjproject 2.5.5 with : ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr Compiled Asterisk 13 with ./configure --libdir=/usr/lib64 All pjproject modules are selectable in menuselect, so here no problem.

So starting up asterisk-11.5.1-2.fc19.x86_64.rpm I get: [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'chan_mgcp.so': /usr/lib64/asterisk/modules/chan_mgcp.so: undefined symbol: ast_pktccops_gate_alloc [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'chan_iax2.so':

Hello, I recently completed a fresh install of Asterisk SVN-group-srtp-r183146M-/trunk , and I'm running into an issue getting the voicemail application module to load. Output from debug shows: ------------------------------- [Aug 11 22:00:01] NOTICE[20173]: loader.c:875 load_modules: 1 modules will be loaded. [Aug 11 22:00:01] WARNING[20173]: loader.c:376 load_dynamic_module: Error loading

Hi all I don't know what went wrong but i no longer seem to be able to compile asterisk. I first do : cd /usr/src/dahdi-linux-2.1.0.4 make clean ; make all ; make install cd /usr/src/dahdi-tools-2.1.0.2 ./configure ; make clean ; make all ; make install ; make config So far.... so good but then when i do : cd /usr/src/asterisk-1.6.0.6 make clean ; ./configure ; make all ; make install i

hi, I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz on Fedora3 linux-2.6.12-1.1372_FC3). It works fine for playlist.ogg from the other CPU, such as 'xmms http://192.168.0.3:8000/listplay.ogg'. But when I use 'stdinpcm' like 'asterisk-ices.xml' which send voip's voice udp packets to 'asterisk-ices.xml' such as; .......(snip)...... <stream>

I'm having trouble with ExternalIVR's socket connection. Is it working in the current 1.6.0 trunk? I'm getting this error: Executing [s at ck987_externIVR:1] ExternalIVR("SIP/itp_jnctn-0000006c", "ivr://127.0.0.1:9012") in new stack [Feb 21 13:38:12] NOTICE[15864]: app_externalivr.c:631 eivr_comm: SIP/itp_jnctn-0000006c: stderr: Failed to execute

Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

I was trying to enable CDR in a mysql-database when the following occured : asterisk*CLI> cdr status CDR logging: enabled CDR mode: simple CDR output unanswered calls: yes CDR registered backend: cdr_manager CDR registered backend: cdr-custom asterisk*CLI> exit [Jun 29 21:56:52] Executing last minute cleanups [root at asterisk asterisk]# vi modules.conf [root at asterisk asterisk]# service

Hello List, To whom it might concern: I have been working in some SlackBuilds (script for making Slackware Packages) for my personal use, but thought they might be useful for someone else here. Beside of the exceptional distributions used so far (CentOS, Debian, Ubuntu, etc.), you might want to test Asterisk on a Slackware Linux box, as it offers outstanding stability and flexibility as

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give