similar to: come back ring

Hi all , In my pbx ,there is no zaptel card ,so i loading ztdummy,but problem appear,when i dial the number into the pbx,sometimes i can not listen to the ivr ,and no rtp create. if i unload the ztdummy driver,the proble will disapper. I guess may be the timer source problem, but i dont't know how to solve it . anyone can give me some advices will be appreciated. asteirsk-1.4.21 and

Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid:

hi everybody, Recently,I did a stress test use asterisk . when I concurrent 100 calls to the queue , i found the waiter need very a long time can listen to the music,about 40s and this time, there are about 30 waiters in the queue. everybody can tell me this reasons.whether the queue's ability is limit or the music on hold concurrent is limit . Thanks in advace! -- Best regards!

G'Day All, How do I fix this: I receive a call at the extension. Press the hold button. Music on hold starts. When I place the handset back on the cradle, the call gets hung up/disconnected. The Phone is A GrandStream Budge Tone 100. Thanks

Hello, I'm trying to use the package ttda, wich is involved in text analysis, for my own data about answers in a company survey. I've installed it, as well as ispell, but when trying to use an example: > zz <- file("stupid.txt", "w") # build a data file > cat("{comment - stupid data file} \n" , file = zz) > cat("<uci=1>

Hello sir, I'm having some problems understading how llvm works. I'm following a concrete example and visualizing the DAG and the .s output and some stuff doesn´t seem to match. I sorry to bother you but I don´t know where I can expose my doubts and I wonder if you could redirect me to some discussion place where someone might help me. Thanks, Patosga

Hi guys, I'm looking for a affordable conference phone and a wifi phone that has a cradle. Polycom seems to make pretty nice conf phones but the price is a bit crazy for us. I saw the recommendation with ATA plus an analog Polycom phone but I do prefer a SIP phone. All because it's just too difficult to pull a phone cable into the current conference room. Is there any cheaper SIP

Dear R fans ( and wiki fans), I am just writing a draft to introduce confidence intervals of various "effect sizes" to my students. Surely, I'll recommend the package MBESS in R. Currently, it means I have to recommend R's interface at first. As a statistics teacher in a dept of psychology, I often have to reply why not to teach SPSS. Psychologists and their students hate to

I read the feature list of asterisk and I cannot see if it is possible to conference a call between extensions. Is it a supported feature of asterisk or is it done in the UA (ATA186 in my case) Here is what I try to do. phone-a -dial-> phone-b tap the cradle (flash on phone-a) phone-a -dial-> phone-c tap the cradle (flash on phone-a) Now I like all 3 phones in a conference call.

I was able to build Samba 3.0.x under Solaris 10 using the Sun Studio compiler (cc) and the Solaris ld (/usr/ccs/bin/ld) that it uses. Trying to build Samba 3.2.0, however, I immediately run into a problem where the build process assumes the linker understands "--version-script", which is specific to the GNU linker: Linking shared library bin/libtalloc.so /usr/ccs/bin/ld: illegal

I need to make a Asterisk 18.0's offline compiling, SVN mp3 support sources downloading does't particulary works cause my asterisk is in an isolated network with NO network access whatsoever, I ve read this thread ( http://lists.digium.com/pipermail/asterisk-users/2013-June/279298.html) but I 'm not understading one thing, because I download the file and run the script but there is no

With Samba 3.0.x, I would: 1. configure 2. make 3. make test 4. make install (if all went well) When I do a 'make test' with Samba 3.2.0, though, I eventually run into problems like this: RUNNING TESTS testparm_s3 Testing netbios name (0) TEST OUTPUT: Load smb config files from /home/david/build/samba-3.2.0/source/st/lib/smb.conf.tmp params.c:Parameter() - Ignoring badly formed line in

Sorry, glad I'm in the right place. Before I start, I want to state that I'm a beginer and I'm trying to develop a backend by adapting an existent target to my platform. My first doubt is about the SelectionDAG and the TargetLowering class. When I use, for example: setOperationAction(ISD::ADD, MVT::i1, Promote); Is it correct to say that I'm promoting any operand used by the

I have a quick question. I did check the docs before asking this but couldn't find an answer to my question. When I first started playing around with winbind I came to the understading from reading one of the docs that winbind needs to be started AFTER smbd and then nmbd in that order. However the rpms I can build from the site which created a winbind service file in /etc/rc.d/init.d have

Hi Running racoon on a Freebsd-4.11 machine gives a kernel panic. I am using the racoon from ports directory which comes with the freebsd installation. Steps followed are as shown below: racoon -f /usr/local/etc/racoon/raccon.conf setkey -f ipsec.conf ping -c 1 <ip_of_the_other_gw> The ping will lead into a crash. The crash dump looks like for th ping packet it

I'm trying to build and install Samba 3.0.30 on a Solaris 9 SPARC machine. When I do a "make test", the RW1 test is failing. If I go back and configure and build 3.0.28 with the same settings, and do a make test, everything passes. Here's what I'm seeing with 3.0.30: ---8<--- Testing RW1 (0) TEST OUTPUT: host=127.0.0.2 share=tmp user=root myname=cannes Running RW1

Environment: Sun Solaris 9 sparc Software: Samba-3.3.3, KRB5-1.6.3, OpenLDAP-2.4.11 Problem: Am trying to create shares with Samba so that users can map to folders on this server using Active Directory. I am successful in creating a Kerberos ticket; I can join the domain; and wbinfo -u and -g give me users in the AD. However, getent passwd only gives me a list of users on the server and not in the

Hi, Just got some CISCO 7960 phones. They seem to work great except if I make any SIP call using the speaker phone (leaving the hand set in the cradle)the call will disconnect in about 5 or so seconds. If I pick up the hand set and make a call, it's fine. Has anyone else run into this ? Any solution ? The phone is on SIP v6.1 - it did the same thing on 4.4 5.0 and 6.0. Thank you in

While pondering several issues, poor quality PSTN POTS lines, potential cost savings with multiple cell numbers, the FAX problems over TDM400p, etc, I wondered about: Cell phone "Base stations" to replace POTS lines. Devices to "cradle" cell phones and connect to TDM400p, for instance, to mimic PSTN. Are there such beasts, how do they play with asterisk? Will FAX work over

I have a Centos 3 server running Samba 3.0.28. It's a member of an AD domain on a Windows Server 2003 R2 Standard x64 SP2 box. From the W2K3 server I can see the samba share I created. Using the Security tab in the Windows Explorer file properties dialog I can add and remove users and change their permissions. However, in the Permissions tab of the Advanced Security Settings dialog,