Software :: Asterisk - Listen To Recorded Call From Phone - Not From Web Interface?
Apr 20, 2011
Maybe I'm not using the right search terms, but I can't find anything on this. I have a setup of Slackware with Asterisk and FreePBX. I am recording calls on demand and can get to the recordings from the Call Monitor (web interface). I would like to be able to access the recording from the phone (Aastra 57i), much like a voice mail is accessed.
Today my boss come to me and ask me to get a cote to upgrade our old "Cisco call manager" (2004) now "Cisco Unified Communications Manager". So I was wondering, instead of doing a costly upgrade (over 35 000�), maybe it's time to change... Does anyone of you got some insight with Asterisk in an enterprise environment? Is it reliable? Following you own judgement, what are the + and - ? If Asterix worth it, what argument (apart of the price) could I use to help the management turning on my side? Will the Cisco 7921 VOIP phone will be able to connect to it? (as we do have over 35 of them)
Enterprise environment:
- 3 sites (VPN interconnected) - ~35 VOIP phones and ~10 landlines phones
I have tryed for hours trying to get this right but failed.I configure the dhcp file and when accepting the changes i get the error NOT CONFIGURED TO LISTEN ON ANY INTERFACE.
My file looks like this
Heres my current settings
I am getting an error: NOT CONFIGURED TO LISTEN ON ANY INTERFACE
I would like to experiment with voip, is there any application which works successfully to make call from PC to phone, other than skype. If its beta also let me know, so that I can contribute my ideas to it. Any resource?
Is there an app out there that will at least mute my speakers or pause my music when I receive a phone call? This would come in handy since I listen to music a bit loud and have a phone that vibrates like a kitten.
I compiled and installed the Asterisk ztdummy package because there is no rpm for it, unfortunately, and i even reinstalled asterisk, but i still get the "No application 'Meetme' for extension..." error when trying to conference. I do a "module show", and it lists other modules that were compiled with the zt source, but not ztdummy.Does anyone know how to fix this? This is more than a passing interest or hobby, because i need to conference about 3 to 5 people to help me test a new Website Content Management System and User Forums Management System i am about to launch as a service.
In practice I have a script that call a java program that call a linux system command. The script if I run it, from a shell functions well,so it is not a java problem. The problem come out when i put this script in a crontab schedulation. The result in this case is that java do not execute the system command. I think it depends on crontab
It's installed as a mobile broadband connection. Only works if my phone is plugged in during boot, otherwise plugging in my phone does nothing. For example, I booted my netbook earlier today but my phone wasn't plugged in. Tethering did not work, it just acts as if it isn't present. I rebooted (with my phone still attached) and now tethering magically works.
I have a mobile phone (T68i) attached to my server and everything works fine. But sometimes the phone craches and needs to be rebooted wich is quite annoying.So I want to use a newer phone that is not 10+ years old :-)But when using a K610i or C702 the phone does not appear on /dev/ttyUSB0 as expected.When inserting the usb cable a dmesg gives this:
Code: [5924924.451033] usb 1-3: new high speed USB device using ehci_hcd and address 19 [5924924.576728] usb 1-3: New USB device found, idVendor=0fce, idProduct=d0d9
I'm using OpenSuse 11.2 64 bits When I try to listen to music in Grooveshark Grooveshark - Listen to Free Music Online - Internet Radio - Free MP3 Streaming I can listen fine, and it seems to work ok, when suddenly the sound of the website stops to work, my processor gets overload and I've to reopen the site to continue to listen. I get the following message in kernel (I'm no sure if it's related)
I recently switched to Ubuntu and I'm loving it, except i am having sound problems with both my laptop and my desktop.My laptop is using an internal microphone (it is a Toshiba Satellite P500) but the mic is not recognized, I hook up my M-Audio USB audio interface which has my studio mic connected to it, it gets recognized as an input device but it doesnt recordSame thing on my desktop, it gets recognized, doesnt record.My laptop is 64-bit and my desktop is 32-bit, both have Ubuntu 10.04
Using Debian Lenny I recorded a wav file with Audacity to burn to a cd with K3b.The problem is when I played the cd with KsCD; it wouldn't stop when it got to the end; it just kept studdering. I didn't have problems using KsCD to play CDs I had ripped and then burned with K3b.Is this a problem with KsCD, or do I need to do something when I finish the file with Audacity?
I recorded some clips on the internet through the sound card withAudacity. I exported the recording as a wave file, and burnt it to a CD. The problem is stop playing. It comes to the end and just repeats the last fraction of a second over and over.I used K3b to burn other audio CDs that I copied and didn't havea problem with KsCD playing them. Is this a problem with Audacity,KsCD or did I do something wrong?I recorded the file; saved project and then exported as a wave file
I have a dell precision m4300 laptop with a 360 wireless bluetooth dell adapter On my system there is a debian lenny with kde3 and backported enable(everythings is p to date except bluez-utils and bluetooth holded at version 3.36-3)
Nowadays bluetooth more less works fine, I can send and receive single file to/from my phone (nokia n70)
The hell begin when I try to browse my phone files from konqueror...with bluetooth:/ I can see the list of all the device near me with sdp://[address]/ i can see two icons (obex file transfer & obex object push) but I cant see or access to any file or folder into the phone.
I also try to update my bluez-utils to 4.60-1~bpo50+1 but in this case kbluetooth totally fail and a see the contextual menu of the system tray icon all disaled.
For about 3-4 weeks, my file system used space was growing and growing. After some days, I decided to analyze the file system in order to understand what is going on. Well, the results returned that the log files at /var/log ( especially kern.log, syslog and messages ) were the log files that were sucking the free space. I searched if this was some kind of bug but it turned out that if these files are growing, the problem is in the records.
So I picked from the tail of the messages log the last 50 lines and I saw this:
I'm running Ubuntu 10.10 in a Samsung R510. I've tried to record sound with an external microphone and there have always been a permanent noise accompanying it.I thought it could be the internal microphone, that's causing the noise, but even when I disabled it the noise remains. Or maybe I didn't disable it the right way !
I am on 64-bit Natty and for the first time since my install a month ago, I have had a system hang requiring a power-off restart. It happened yesterday and a few moments ago.My question is that I want to troubleshoot this myself and wonder if the activities that my system were running just prior to crashing are recorded in a log file somewhere.I used to look int /var/log.messages or /var/log/boot, but this was in the Maverick days.
I want to measure the time it takes to execute a C function. The question is which is the best method to record time. After a lot of searching online, I found two ways [URL]:
a. using clock_gettime(CLOCK_MONOTONIC, &t) b. using TSC
I call the C function 50 times (recording the time it takes for each call). The average is found only for the last 35 calls.
Using (a), I get:
48345 us 28350 us 28379 us 27716 us
[code]....
Average time for the last 35 loops: 121450.47 microseconds
Average for the last 35 loops: 27236.66 cycles What could be causing the strange value of 3322259 us in method (a)? I could just use method (b) but I would like to know what is going on here... Btw, I am using a desktop - with Linux debian 2.6.26 #1 SMP.
I have recently installed OpenSUSE 11.3. I can play audio files, but there is a problem recording, namely the recording is way too slow. This was tried 3 ways: in Audacity, in Skype (making a test call), and using the arecord command in a console, as root. To make the playback sound normal, I tried taking the already-recorded file and speeding it up in Audacity. It was necessary to speed up the recording by 50%. I've tried updating alsa and other suggestions which I could figure out, and run the scripts for diagnostic information. BTW, I'm using a headset plugged in to standard 1/8 inch mic and speaker jacks, not a USB headset.
I tried ubuntu 10.0 yesterday (running from my usb pen) and I liked it. A lot. However, there are a few issues that i need to know if there's any way to solve them, before making the transition: Keyboard keys are swapped; I mean, if I press alt+(anyNumber) I get another character. Why does this happen?-I develop a lot of code in VBA; I tried to record a macro using Open Office Calc, but I got the impression that the code used in the recorded macro is slightly different.
I am trying to record using a microphone on a machine running Fedora 14. The microphone itself seems to work fine when plugged into a Windows laptop, and the Linux machine is able to play sound just fine. However, although I can record using the microphone (using arecord or audacity, for example), the recorded audio is super, super quiet. I have run alsamixer, and experimented with every capture source. Anything which had any effect (using arecord -vv to see the dynamic sound level) has been turned up to 100%. Still, recorded audio is barely audible.
I have run: yum reinstall alsa-utils alsa-plugins-pulseaudio alsa-tools-firmware alsa-firmware alsa-lib But that made no difference. I ran alsa-info, and the output is here: [URL]. Why my recorded sound is barely audible? My next step is probably going to be to boot to Ubuntu, just to see if it has the same problem.
I am having some problems with recording with my laptop. When I try to record something through the laptops mic, usb webcam mic or with desktop recorders audio record feature all playback is too fast (sounds a bit like alvin and the chipmunks :L ) Is anybody else having this problem, or even better know how to fix it ?
Sorry if this has been asked before, I'm running 10.10 and had never tried recording before I upgraded to 10.04. I had the same problem in 10.04 though.
My wife is taking an online language course that requires Windows in order to record and playback speech (for pronunciation training). I decided the easiest thing to do would be to clone a VM I use on my machine for tasks that can only be accomplished in a Windows environment. I did so, and successfully installed the speech recognition utility on the Windows guest VM. Unfortunately, although the microphone works just fine--I can hear my voice in the headphones--no sound is recorded by the speech recognition utility. Just to be clear, the microphone is fully functional under GNU/Linux. My wife has been using Skype without any issues at all. That is, Skype used to work. Somewhere during the course of trying to resolve the voice recording problem I managed to break Skype. I have no idea how.
It's a long story, but I'll try to be as brief as possible. When I could not get the VM solution to work properly, I thought maybe the problem was with the VM, so I checked the original copy on my own machine. No problem there. Windows sees and hears the microphone just fine (using the testing utility under sound configuration). So how is it possible that the microphone doesn't work in the cloned VM image? Before you stop me, allow me to note that both machines have identical motherboards, and both are configured to use the onboard sound. Sound works on the Linux host and the Windows guest VM. Both sound and microphone work in the original VM image on my machine, but only sound works in the cloned VM on my wife's machine.
If you've followed me down the rabbit hole this far, please continue with me a bit further as things get stranger and stranger. After triple checking all the setting in KMix (I'm running Squeeze/KDE on both machines) to be sure they were identical, I noted that I was unable to view/add/enable the "channel" option on my wife's machine. Why, I have no idea. As I said, the machines are identical as can be both regarding hardware and software. Out of desperation, I decided to set up an account for my wife on my machine. I created her account, made sure she had the necessary audio permissions, then cloned my VM to her account. Same problem.
I would love for someone to point out what I'm missing here. Same hardware, identical VM (clone), same permissions, same sound configurations in KMix and under the Windows guest VM. How is it possible for the microphone to work for me, but not for my wife? Is there some mystery configuration file somewhere that has magically been altered for her account alone? The worst part about this entire process is that I've broken my wife's Skype, which is a big deal seeing as we live in a tiny country in the heart of Africa and cell phone communications are very expensive.
I have a lot of recordings on my mythtv box.i can see/play them fine within the mythfrontend interface, and mythweb loads almost all pages ok (including schedules, guides, etc.), but when I click on 'recorded programs' at the top, Allowed memory size of 33554432 bytes exhausted (tried to allocate 88 bytes) in /usr/ share/ mythtv/mythweb/includes/translate.php on line 142.I've had this error before, and it went away when I purged my list of recordings.
I'm trying to record video off my webcam, and tried two different programs - cheese and wxcam.
I honestly didn't have much luck with cheese - it seemed to work fine, but when it came time to record video, the frame rate dropped to a ridiculous level, making the video completely unusable. I could wave my arm in front of the camera, and it might pick up one frame of that.
Moving on, I tried wxcam, which I like a lot.. however, when I record video with the xvid compression, they play back considerably faster than they were recorded.
I have a USB 3g modem, which also supports voice calling, I want to implement IVR ( Interactive Voice response) with it on linux, Can I do it with Asterisk? If yes, then please help me on how to do this ( references) ?
Is there example or howtos available to do this?
Can I develop IVR system in linux without Asterisk?