similar to: problems compiling shadydial-asterisk on gentoo

Hi All, Somebody has installed shadydial?, shadidial is used for predictive dialers, I need to know the following: when a call is answered, this call es sending a to agent, but I need that the agent when receive the call in the desktop appear all information of this number. Somebody have experience in this type of system of remote agent. Regards, Note: Sorry for my bad english.

When I make, I obtain the following errors Using FLAGS = -O -D_SAMBA_BUILD_ -I./popt -Iinclude -I/usr/local/src/samba-3.0. 21c/source/include -I/usr/local/src/samba-3.0.21c/source/ubiqx -I/usr/local/src/ samba-3.0.21c/source/tdb -I. -D_LARGEFILE64_SOURCE -D_FILE_OFFSET_BITS=64 -D_GN U_SOURCE -I/usr/local/src/samba-3.0.21c/source -D_SAMBA_BUILD_ LIBS = -lcrypt -lresolv -lnsl -ldl

I successfully built llvm using msys/mingw, but I am getting the following error when I try to build llvm-gcc using $ ../llvm-gcc4.2-2.6.source/configure --prefix=/c/Temp/msys-llvm-gcc --program- prefix=llvm- --enable-llvm=/c/Temp/msys-llvm --enable-languages=c,c++ --disable -shared $make -------------------------------------------------- gcc -c -g -DIN_GCC -W -Wall -Wwrite-strings

When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help

Hi to all, I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib/libc.so.6 #2 0x005b5631 in abort () from /lib/libc.so.6 #3 0x005ebe6b in __libc_message () from /lib/libc.so.6 #4 0x005f3b16 in _int_free ()

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\"

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

O -fPIC -c -o chan_agent.o chan_agent.c chan_agent.c: In function `login_exec': chan_agent.c:595: parse error before '<<' token chan_agent.c:602: parse error before '>>' token make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels'

Hi, I have 1.2.9.1 installed. It always rings first available agents for 15 seconds, then rings and hangs up the next agents straight away, then ring the next agents for 15 seconds. It goes as a loop. Any one has the following same problem? Thanks. Agents.conf [general] persistentagents=yes [agents] autologoff=60 wrapuptime=15000 ackcall=no group=1 agent => 7130,7130,agent1 agent =>

after recompilling asterisk (trunk-r75109) after system (mandriva cooker) update (new glibc 2.6, gcc 4.2.1), sound starts very choppy, when codec translation is performed, if translation isn't needed, it sounds OK any idea? until update, everything worked fine. I'm using ztdummy as clock source. during compile, I got lot of errors... ael_main.c: In function ?ast_context_add_ignorepat2?:

Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.

chan_agent.c: In function `login_exec': chan_agent.c:595: parse error before '<<' token chan_agent.c:602: parse error before '>>' token make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1

Hi, am building the latest version of Asterisk (1.4.13) on a self-build Linux host (based on LFS-6.3). Version 1.4.12 built installed and worked fine. Last night I upgraded the kernel to 2.6.23 and rebuilt the zaptel driver package 1.4.5.1 against it. That seemed to build and install O.K. too. I dl'd the 1.4.13 source tarball and tried to build that: ./configure ran O.k. 'make

Hi all, Have been experimenting with agents and queues instead of placing calls direct to a user's phone extension, but I've run into problems with calls to both the agent and the extension which creates a zombie and double records calls abandoned etc. We're using a unique queue for each agent (only a handful of users) to try and get some agent/queue information to see what the

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

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

If I have in my dialplan [AgentQ] exten => _XX.,1,Dial(Sip/{$exten},120,g) exten => _XX.,2,NoOP(here we are) where [AgentQ] is called by the queue command to a member added by addqueuemember(Local/99@AgentQ) why don't I get to the NoOp if the agent hangs up during the announcement message (to the agent) ? I see in the app_dial.c program that the "g" flag is tested thus:

I was wanting to use the new MuxMon application to record calls - it seems to be a "nicer" way of recording than the Monitor application. However, there is a slight issue with agents - we use the recordcalls option in agents.conf to record inbound agent calls - and I believe from looking at the source code that is uses the monitor application. Is there any way to get chan_agent to

Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on "pause". I'm using chan_agent for the agents, so agents are

I am beating my head against a problem where queue calls offered by Agent channel to a SIP UA cannot be REFER transferred if the target UA/extension hasn't accepted the call. If the members of the queue are SIP channels, this is not a problem. I suspect chan_agent isn't flagging the bridge from Zap/n -> SIP/n properly, or this is by design. The following line is what is spoken before