similar to: Hangup detection with TDM400 in UK

Hi I'm presently installing 2.2.8a on Mandrake 9.1 I'm a newbie to Linux as well as Samba, so it's been fun... When I first installed Linux I installed Samba from the discs (2.2.7a), but on reading up on the subject it became clear that I should give installing the source a go. My problem is this - I don't think I have rerouted all of the paths to usr/local/samba from what

I have an analogue line connected to an FXO module on a TDM400. I want to use the FXO line only for outgoing calls - incoming calls are answered by other equipment, but even if I don't specify an "s" extension, the FXO module still seizes the line for all incoming calls. Is it possible to configure the FXO module to ignore ring events completely? Wiki resources etc have been no

Dear knowledgeable experts :-), I am trying to get openmpi, Rmpi and SNOW running on a Myrinet/GM cluster. I'm not an IT expert, but I surely could use a working installation of Rmpi and SNOW. I try to load the Rmpi library and get the following: > library(Rmpi) [master:07230] mca: base: component_find: unable to open osc pt2pt: file not found (ignored)

I've followed the recent thread on caller id with UK British Telecom networks (where the caller id data is delivered before the first ring). My understanding is that if I use a recent CVS head (e.g. CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to do is include the line: usecallerid=uk In my zapata.conf (in the [channels] section) I've done this, but I get: Sep

I am running a fully updated Fedora Core 3 server, and installed a pretty thin system, and have just installed packages as needed. My problem is that I am trying to get asterisk-perl installed, but it keeps segmentation faulting on me. I know a little python but perl baffles me. # perl Makefile.PL Segmentation fault ==========Strace============

i chose to install tomcat during the OS install as well as java....however, since I needed to use a different jvm, i used the method described in http://wiki.centos.org/HowTos/JavaOnCentOS for Sun jdk 1.5.0_13 and it worked fine as I can tell, as using the alternatives --config java command, i see both 1.4.2 and 1.5.0 versions and choosing the 1.5.0 version yields the right version when doing a

Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -------------- next part -------------- An HTML attachment was

I'm considering purchasing a Digium TDM400 card with FXO module to connect my asterisk box up to BT's PSTN. I've searched Google and have found that in the past there have been problems with this card detecting the calling party hangup. I am interested to know if this issue has now been resolved and if UK caller ID is now fully supported. Also if anybody using the card in the UK is

So I have an application where the users want to divert incoming calls on one analogue line out to another analogue line - both lines are supplied by BT and theres a TDM400 in the box. Call comes in, system Dial's the forwarding number and bridges the calls. Works fine. Until one (or both) leg hangs up. And the box never sees the hangup. So it sits there with both Zap channels open

And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of

Hello All, I am using monitor with soxmix, however the quality seems somewhat low after sox converts to mp3. Does anyone know a way to get a higher quality file? Some of my lines are coming in on isdn. Regards, Greg

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a command to run after a queue call finishes. Any hints?

Good day all I'm getting this error while trying to dial out on my asterisk server using a openline4 card "exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872" Please Help me

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

Hello, Hacker I install VoiceTronix OpenSwitch 12 port PCI Telephone Card, and setting vpb.conf, extensions.conf following My problem is: When i dial to fxo(channel 9-12), it is ok, but when i continue press exten 102, the channel crach with error messages following exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872 Do i ignore some setting for VoiceTronix OpenSwitch12

> Hi Daniel, > > I'm intrigued by this and wanted to try it out - but I'm wondering how > you get Asterisk to call sox at all during Voicemail()? Our server > doesn't even have sox installed, so I'm not sure how to go about > tricking Asterisk into running a different one. To do anything useful you would have to get sox installed on your server. But to get

Hi all, I'm using ConfBridge within Asterisk 1.6.20 and want to record the conference, so I'd like to start the recording when the second user joins, so in the example below, for example, how can I get the current user count in ConfBridge 3000? [conferences] ;authenticated conference (ext C-O-N-F = 2663) exten => 2663,1,Answer same => n,Wait(1) same => n,Authenticate(143382)

Hi, I've just noticed something strange in dovecot 0.99.10.5. I created some IMAP folders from the command line (mkdir .INBOX.Folder1) as a different user to the owner of the mailbox. Obviously when I tried to view them via IMAP I got an error message, but the message to the client was "internal error[date time]" where date and time appeared to be the current date and time. Is