Re: [RDD] Voip Based Call Manager
On Apr 22, 2013, at 12:07 10, Bill Putney wrote: > I keep encouraging Fred Gleason to spend some time on CallCommander. > That could be a really cool VoIP/AoIP broadcast call manager. FWIW, there is experimental Asterisk support in CallCommander. http://www.rivendellaudio.org/ftpdocs/callcommander/ Cheers! |-| | Frederick F. Gleason, Jr. | Chief Developer | | | Paravel Systems | |-| | A room without books is like a body without a soul.| | -- Cicero | |-| ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Yes agreed, Apologies to everyone, hope it has helped though :) -Original Message- From: rivendell-dev-boun...@lists.rivendellaudio.org [mailto:rivendell-dev-boun...@lists.rivendellaudio.org] On Behalf Of sas...@radio42.de Sent: Wednesday, 24 April 2013 11:30 PM To: User discussion about the Rivendell Radio Automation System Subject: Re: [RDD] Voip Based Call Manager Hi As this whole conversation is a bit off-topic from rivendell... I would suggest that you all can send me personal messages to sas...@radio42.de and I will add the consent of those messages to the wiki. I think a general how-to is now in the last 10 messages :) And I don't wanna bug this list with more Voip Based Call Manager mails ;-) Best regards, Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Hi As this whole conversation is a bit off-topic from rivendell... I would suggest that you all can send me personal messages to sas...@radio42.de and I will add the consent of those messages to the wiki. I think a general how-to is now in the last 10 messages :) And I don't wanna bug this list with more Voip Based Call Manager mails ;-) Best regards, Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Am 2013-04-24 15:15, schrieb Lee Baker: Hi Sascha, Have just been following your wiki instructions, when I run ./ yateconnector.tcl I get this error "/usr/bin/env: tclsh8.5: No such file or directory" perhaps I am missing something? looks like you are missing tcl in version 8.5. You may try to edit the shebang line #!/usr/bin/env tclsh8.5 in yateconnector.tcl to whatever version of tcl is available. But I only tested it with tcl 8.5. Maybe an apt-get install tcl8.5 will help Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Am 2013-04-24 14:47, schrieb Lee Baker: > Oh I see now! > So now for my next question, my radio station has a telephone hybrid > which > uses a voip ATA. > Currently what we have to do to put a caller to air, if a call comes > in on > the studio number it is answered from the studio phone and then the > announcer has to transfer the call to the hybrid extension, pick the > call up > using a normal PSTN phone and then place the call to air. > > Would OAP alleviate the need to do this by handling the transfer from > the > push of a button on the console screen? This could be done like this in the future: Call comes in on the studio number. Call is being shown in OAP Screener. You now have multiple options: 1. take the call with a studio or screener phone (just select the right device on the right) and push answer. This will connect the call to the selected number. If the phone is in auto-answer mode you have the caller directly on the line. 2. take the call with blink or your voip ata. Just create an extension in yate and register your ata with yate. select the device which you configured in studio1.conf on the yateconnector and there you go. Another hint, because it's not obvious: If you click on the line number you will get a details dialog where you can fill in the name, town, or whatever you have screened prior to take the call on air. Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Hi Sascha, Have just been following your wiki instructions, when I run ./ yateconnector.tcl I get this error "/usr/bin/env: tclsh8.5: No such file or directory" perhaps I am missing something? -Original Message- From: rivendell-dev-boun...@lists.rivendellaudio.org [mailto:rivendell-dev-boun...@lists.rivendellaudio.org] On Behalf Of sas...@radio42.de Sent: Wednesday, 24 April 2013 10:40 PM To: User discussion about the Rivendell Radio Automation System Subject: Re: [RDD] Voip Based Call Manager Am 2013-04-24 14:26, schrieb Lee Baker: > Sorry one other question, how does the audio routing work? > > Can this work with jack? Or does it just use the default sound card > settings? blink can work with alas-jack. I created a special virtual sound card in asound.conf and gave it a decent name. So I can route the input and output of blink to any soundcard or jack-mixer port or whatever. Please keep in mind that OnAirPhone doesn't handle audio at all. It's only the message-routing and multi-user glue code to take/hold and place calls. For example: If you click on "ANSWER" the ringing line gets transferred to the selected SIP-Phone or to the number which is the blink client. In my test setup I use some Cisco 7960 Phones for the screener desks and the blink client for the 'on-air' stuff. I also configured audacity to record the talents mic and the caller in a special talkback mode. So you can record calls of air with blink on seperate channels. It's basically a big switching and re-routing in jack. Then I have a audacity macro which exports the file into a directory. The on air talent just has to press the audacity macro hotkey and it will export. The export dir is observed via python inotify. When a file arrives it is beeing renamed to something recognizable like "PhoneCall_2013-04-24_14-35.wav" and then imported into rivendell. With nearly no latency. You have the file available in RD in just 1-2 seconds after you start the export macro. I know there might be questions on the how to to such a script... :) I will also post that on the wiki :) Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Oh I see now! So now for my next question, my radio station has a telephone hybrid which uses a voip ATA. Currently what we have to do to put a caller to air, if a call comes in on the studio number it is answered from the studio phone and then the announcer has to transfer the call to the hybrid extension, pick the call up using a normal PSTN phone and then place the call to air. Would OAP alleviate the need to do this by handling the transfer from the push of a button on the console screen? -Original Message- From: rivendell-dev-boun...@lists.rivendellaudio.org [mailto:rivendell-dev-boun...@lists.rivendellaudio.org] On Behalf Of sas...@radio42.de Sent: Wednesday, 24 April 2013 10:40 PM To: User discussion about the Rivendell Radio Automation System Subject: Re: [RDD] Voip Based Call Manager Am 2013-04-24 14:26, schrieb Lee Baker: > Sorry one other question, how does the audio routing work? > > Can this work with jack? Or does it just use the default sound card > settings? blink can work with alas-jack. I created a special virtual sound card in asound.conf and gave it a decent name. So I can route the input and output of blink to any soundcard or jack-mixer port or whatever. Please keep in mind that OnAirPhone doesn't handle audio at all. It's only the message-routing and multi-user glue code to take/hold and place calls. For example: If you click on "ANSWER" the ringing line gets transferred to the selected SIP-Phone or to the number which is the blink client. In my test setup I use some Cisco 7960 Phones for the screener desks and the blink client for the 'on-air' stuff. I also configured audacity to record the talents mic and the caller in a special talkback mode. So you can record calls of air with blink on seperate channels. It's basically a big switching and re-routing in jack. Then I have a audacity macro which exports the file into a directory. The on air talent just has to press the audacity macro hotkey and it will export. The export dir is observed via python inotify. When a file arrives it is beeing renamed to something recognizable like "PhoneCall_2013-04-24_14-35.wav" and then imported into rivendell. With nearly no latency. You have the file available in RD in just 1-2 seconds after you start the export macro. I know there might be questions on the how to to such a script... :) I will also post that on the wiki :) Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Am 2013-04-24 14:26, schrieb Lee Baker: > Sorry one other question, how does the audio routing work? > > Can this work with jack? Or does it just use the default sound card > settings? blink can work with alas-jack. I created a special virtual sound card in asound.conf and gave it a decent name. So I can route the input and output of blink to any soundcard or jack-mixer port or whatever. Please keep in mind that OnAirPhone doesn't handle audio at all. It's only the message-routing and multi-user glue code to take/hold and place calls. For example: If you click on "ANSWER" the ringing line gets transferred to the selected SIP-Phone or to the number which is the blink client. In my test setup I use some Cisco 7960 Phones for the screener desks and the blink client for the 'on-air' stuff. I also configured audacity to record the talents mic and the caller in a special talkback mode. So you can record calls of air with blink on seperate channels. It's basically a big switching and re-routing in jack. Then I have a audacity macro which exports the file into a directory. The on air talent just has to press the audacity macro hotkey and it will export. The export dir is observed via python inotify. When a file arrives it is beeing renamed to something recognizable like "PhoneCall_2013-04-24_14-35.wav" and then imported into rivendell. With nearly no latency. You have the file available in RD in just 1-2 seconds after you start the export macro. I know there might be questions on the how to to such a script... :) I will also post that on the wiki :) Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
It's my pleasure to be playing with this, I have searched for so long to find a solution such as this and you have done a fantastic job in putting it together! It's great to have people such as yourself sharing applications like this to the community and putting in such a great effort. I am trying to convert my local community radio station to open source software and get them away from winblow$$$ This will be a fantastic addition to our studios! -Original Message- From: rivendell-dev-boun...@lists.rivendellaudio.org [mailto:rivendell-dev-boun...@lists.rivendellaudio.org] On Behalf Of sas...@radio42.de Sent: Wednesday, 24 April 2013 10:29 PM To: User discussion about the Rivendell Radio Automation System Subject: Re: [RDD] Voip Based Call Manager Am 2013-04-24 14:07, schrieb Lee Baker: > Hi Sascha > Thanks for the below instructions, I successfully "made" everything > and can open the OnAirPhone main screen. > I have installed Yate on the same VM as OAP, I run com_server and get > "2013-04-24 22:00:03 Yate server error: Connection refused" Please use File->Settings to set the correct IP for the OAS Yate Connector. In your case it's 127.0.0.1 (localhost). That should do the trick :) > Thanks once again for your help on this. You are welcome. I really appreciate someone trying the stuff so I can get my not existing documentation up to date. :-) Please stay tuned, I add some yate config snippets to the wiki in a few minutes. Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Am 2013-04-24 14:07, schrieb Lee Baker: > Hi Sascha > Thanks for the below instructions, I successfully "made" everything > and can > open the OnAirPhone main screen. > I have installed Yate on the same VM as OAP, I run com_server and get > "2013-04-24 22:00:03 Yate server error: Connection refused" Please use File->Settings to set the correct IP for the OAS Yate Connector. In your case it's 127.0.0.1 (localhost). That should do the trick :) > Thanks once again for your help on this. You are welcome. I really appreciate someone trying the stuff so I can get my not existing documentation up to date. :-) Please stay tuned, I add some yate config snippets to the wiki in a few minutes. Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Sorry one other question, how does the audio routing work? Can this work with jack? Or does it just use the default sound card settings? Regards, Lee -Original Message- From: rivendell-dev-boun...@lists.rivendellaudio.org [mailto:rivendell-dev-boun...@lists.rivendellaudio.org] On Behalf Of sas...@radio42.de Sent: Wednesday, 24 April 2013 7:07 PM To: User discussion about the Rivendell Radio Automation System Subject: Re: [RDD] Voip Based Call Manager Am 2013-04-23 23:18, schrieb Lee Baker: > Hi Sascha > > That's great! Excited to see a package like this. > > I am running Ubuntu 12.4, I have now setup python, python-qt4, git, > yate, yateadmin and blink. > > I have downloaded your package via git, just wondering if you might be > able to post some documentation on how to run and configure. > > I go to run the oapscreener.py and get the below message > > onair@onair-VirtualBox:~/Downloads/OnAirPhone$ ./oapscreener.py > Traceback (most recent call last): > File "./oapscreener.py", line 45, in > from screener import Ui_MainWindow > ImportError: No module named screenerscreen Perhaps I am doing > something incorrectly? > Cheers, > Lee you have to call "make" so that all the python resource and ui files are built :-) but that's only needed once in a while when you make changes to the qt-resource or ui files. The makefile uses the tool "pyuic" to generate the python ui files from the Qt .ui files. In the git tree there is a subdir called "yate" which has a sample config "studio1.conf" and the yateconnector.tcl. IMHO you can just run the yateconnector.tcl on the machine where yate is running. The yateconnector should connect to the yate port 5039 which may be enabled in the yate configfile (extmodule.conf). If you have a connection to yate it should give you some output like this: debian:~/OnAirPhone/yate$ ./yateconnector.tcl Yate OAP Server 0.1 * reading config - Service Name:Test Radio - Phone Lines: 6 - Handled Number: 111 "Local Call" - Handled Number: 628 "Studio Hotline" - Line Mode: numeric - Device: 100 OnAir Console - Device: 101 Studio 1 - Device: 102 Screener 1 - Device: 105 Test Client * ready Then you have to start the com_server.py on a machine within an ipv4-routable network :-) so I guess in your case it's on the same virtualmachine :-) com_server.py reads it's config directly from the yateconnector.tcl. So the basic idea is: yate <-> yateconnector.tcl <-> com_server.py <-> multiple instances of oapscreener.py on multiple PCs Hope that get's the first run done :-) I will also add this mail to my wiki located at http://www.astrastudio.de/wiki/ for further reference. I will also add yate regexroute.conf config snippets so you know how to setup the 'incoming' "Handled Numbers" in yate. Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Hi Sascha Thanks for the below instructions, I successfully "made" everything and can open the OnAirPhone main screen. I have installed Yate on the same VM as OAP, I run com_server and get "2013-04-24 22:00:03 Yate server error: Connection refused" Just a quick ref for anyone else trying to run "make" you have to first make sure you have pyqt4-dev-tools installed. Back to Yate and the connection being refused, do I have to configure something else to make OAP talk to Yate? Thanks once again for your help on this. Kind Regards, Lee -Original Message- From: rivendell-dev-boun...@lists.rivendellaudio.org [mailto:rivendell-dev-boun...@lists.rivendellaudio.org] On Behalf Of sas...@radio42.de Sent: Wednesday, 24 April 2013 7:07 PM To: User discussion about the Rivendell Radio Automation System Subject: Re: [RDD] Voip Based Call Manager Am 2013-04-23 23:18, schrieb Lee Baker: > Hi Sascha > > That's great! Excited to see a package like this. > > I am running Ubuntu 12.4, I have now setup python, python-qt4, git, > yate, yateadmin and blink. > > I have downloaded your package via git, just wondering if you might be > able to post some documentation on how to run and configure. > > I go to run the oapscreener.py and get the below message > > onair@onair-VirtualBox:~/Downloads/OnAirPhone$ ./oapscreener.py > Traceback (most recent call last): > File "./oapscreener.py", line 45, in > from screener import Ui_MainWindow > ImportError: No module named screenerscreen Perhaps I am doing > something incorrectly? > Cheers, > Lee you have to call "make" so that all the python resource and ui files are built :-) but that's only needed once in a while when you make changes to the qt-resource or ui files. The makefile uses the tool "pyuic" to generate the python ui files from the Qt .ui files. In the git tree there is a subdir called "yate" which has a sample config "studio1.conf" and the yateconnector.tcl. IMHO you can just run the yateconnector.tcl on the machine where yate is running. The yateconnector should connect to the yate port 5039 which may be enabled in the yate configfile (extmodule.conf). If you have a connection to yate it should give you some output like this: debian:~/OnAirPhone/yate$ ./yateconnector.tcl Yate OAP Server 0.1 * reading config - Service Name:Test Radio - Phone Lines: 6 - Handled Number: 111 "Local Call" - Handled Number: 628 "Studio Hotline" - Line Mode: numeric - Device: 100 OnAir Console - Device: 101 Studio 1 - Device: 102 Screener 1 - Device: 105 Test Client * ready Then you have to start the com_server.py on a machine within an ipv4-routable network :-) so I guess in your case it's on the same virtualmachine :-) com_server.py reads it's config directly from the yateconnector.tcl. So the basic idea is: yate <-> yateconnector.tcl <-> com_server.py <-> multiple instances of oapscreener.py on multiple PCs Hope that get's the first run done :-) I will also add this mail to my wiki located at http://www.astrastudio.de/wiki/ for further reference. I will also add yate regexroute.conf config snippets so you know how to setup the 'incoming' "Handled Numbers" in yate. Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Am 2013-04-23 23:18, schrieb Lee Baker: > Hi Sascha > > That's great! Excited to see a package like this. > > I am running Ubuntu 12.4, I have now setup python, python-qt4, git, > yate, > yateadmin and blink. > > I have downloaded your package via git, just wondering if you might be > able > to post some documentation on how to run and configure. > > I go to run the oapscreener.py and get the below message > > onair@onair-VirtualBox:~/Downloads/OnAirPhone$ ./oapscreener.py > Traceback (most recent call last): > File "./oapscreener.py", line 45, in > from screener import Ui_MainWindow > ImportError: No module named screenerscreen > Perhaps I am doing something incorrectly? > Cheers, > Lee you have to call "make" so that all the python resource and ui files are built :-) but that's only needed once in a while when you make changes to the qt-resource or ui files. The makefile uses the tool "pyuic" to generate the python ui files from the Qt .ui files. In the git tree there is a subdir called "yate" which has a sample config "studio1.conf" and the yateconnector.tcl. IMHO you can just run the yateconnector.tcl on the machine where yate is running. The yateconnector should connect to the yate port 5039 which may be enabled in the yate configfile (extmodule.conf). If you have a connection to yate it should give you some output like this: debian:~/OnAirPhone/yate$ ./yateconnector.tcl Yate OAP Server 0.1 * reading config - Service Name:Test Radio - Phone Lines: 6 - Handled Number: 111 "Local Call" - Handled Number: 628 "Studio Hotline" - Line Mode: numeric - Device: 100 OnAir Console - Device: 101 Studio 1 - Device: 102 Screener 1 - Device: 105 Test Client * ready Then you have to start the com_server.py on a machine within an ipv4-routable network :-) so I guess in your case it's on the same virtualmachine :-) com_server.py reads it's config directly from the yateconnector.tcl. So the basic idea is: yate <-> yateconnector.tcl <-> com_server.py <-> multiple instances of oapscreener.py on multiple PCs Hope that get's the first run done :-) I will also add this mail to my wiki located at http://www.astrastudio.de/wiki/ for further reference. I will also add yate regexroute.conf config snippets so you know how to setup the 'incoming' "Handled Numbers" in yate. Best regards Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Hi Sascha That's great! Excited to see a package like this. I am running Ubuntu 12.4, I have now setup python, python-qt4, git, yate, yateadmin and blink. I have downloaded your package via git, just wondering if you might be able to post some documentation on how to run and configure. I go to run the oapscreener.py and get the below message onair@onair-VirtualBox:~/Downloads/OnAirPhone$ ./oapscreener.py Traceback (most recent call last): File "./oapscreener.py", line 45, in from screener import Ui_MainWindow ImportError: No module named screenerscreen Perhaps I am doing something incorrectly? Cheers, Lee -Original Message- From: rivendell-dev-boun...@lists.rivendellaudio.org [mailto:rivendell-dev-boun...@lists.rivendellaudio.org] On Behalf Of sas...@radio42.de Sent: Wednesday, 24 April 2013 12:07 AM To: User discussion about the Rivendell Radio Automation System Subject: Re: [RDD] Voip Based Call Manager Am 2013-04-23 00:10, schrieb Lee Baker: > Hi Sascha, > That looks quite nice, when do you think you will have it ready for > testing in the community? Would be keen to check it out. > Cheers, > Lee Hi Lee You can check it out right now if you want and if you have no fear to get those running: * python * pyqt * yate / tcl you can check out the git code from here: git clone git://rc5.de/OnAirPhone.git I'm sorry for the fact that it's not very well documented on what todo to get it running. If you have questions please send me an email and I will add some documentation on our wiki about the things that might be in question. Best regards, Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Am 2013-04-23 00:10, schrieb Lee Baker: > Hi Sascha, > That looks quite nice, when do you think you will have it ready for > testing > in the community? Would be keen to check it out. > Cheers, > Lee Hi Lee You can check it out right now if you want and if you have no fear to get those running: * python * pyqt * yate / tcl you can check out the git code from here: git clone git://rc5.de/OnAirPhone.git I'm sorry for the fact that it's not very well documented on what todo to get it running. If you have questions please send me an email and I will add some documentation on our wiki about the things that might be in question. Best regards, Sascha ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Hi Sascha, That looks quite nice, when do you think you will have it ready for testing in the community? Would be keen to check it out. Cheers, Lee -Original Message- From: rivendell-dev-boun...@lists.rivendellaudio.org [mailto:rivendell-dev-boun...@lists.rivendellaudio.org] On Behalf Of sas...@radio42.de Sent: Tuesday, 23 April 2013 12:30 AM To: User discussion about the Rivendell Radio Automation System Subject: Re: [RDD] Voip Based Call Manager Am 2013-04-22 06:55, schrieb Lee Baker: > Hi all, just wondering if anyone knows of any open source voip based > call managers like Phonebox? > Have been trying to find a nice solution for this. > Cheers > Lee Hi Lee Due to the lack of such a system, I started working on one :-) It's called OnAirPhone and is meant to be a complete call/screening solution. But it's really early alpha :) You may have a look at the code here: https://rc5.de/gitweb/?p=OnAirPhone.git;a=summary and here is a small screenshot: http://www.astrastudio.de/wp-content/uploads/2013/02/Screenshot-OnAirPhone-S creener.png You will need yate http://yate.null.ro/pmwiki/ as soft switch and any SIP based phone for the screening people and/or a sip softphone as hybrid replacement (to/from on-air/broadcast console) You may use yate-qt4 (the sip client provided by yate) or as I do: a source-modified version of blink http://icanblink.com/download.phtml so that the client does auto-answer. With this you can completely control everything from within the OnAirPhone Screener module: Take calls, add notes & names, hold calls, transfer calls, hangup calls. Everything from multiple PCs to multiple SIP clients/phones. Even over openvpn. But as I said: It's not completed by now and has still some features missing. Best regards Sascha Ludwig ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
I keep encouraging Fred Gleason to spend some time on CallCommander. That could be a really cool VoIP/AoIP broadcast call manager. Bill On 4/21/13 9:55 PM, Lee Baker wrote: > Hi all, just wondering if anyone knows of any open source voip based call > managers like Phonebox? > > Have been trying to find a nice solution for this. > > Cheers > Lee > > ___ > Rivendell-dev mailing list > Rivendell-dev@lists.rivendellaudio.org > http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
Re: [RDD] Voip Based Call Manager
Am 2013-04-22 06:55, schrieb Lee Baker: > Hi all, just wondering if anyone knows of any open source voip based > call > managers like Phonebox? > Have been trying to find a nice solution for this. > Cheers > Lee Hi Lee Due to the lack of such a system, I started working on one :-) It's called OnAirPhone and is meant to be a complete call/screening solution. But it's really early alpha :) You may have a look at the code here: https://rc5.de/gitweb/?p=OnAirPhone.git;a=summary and here is a small screenshot: http://www.astrastudio.de/wp-content/uploads/2013/02/Screenshot-OnAirPhone-Screener.png You will need yate http://yate.null.ro/pmwiki/ as soft switch and any SIP based phone for the screening people and/or a sip softphone as hybrid replacement (to/from on-air/broadcast console) You may use yate-qt4 (the sip client provided by yate) or as I do: a source-modified version of blink http://icanblink.com/download.phtml so that the client does auto-answer. With this you can completely control everything from within the OnAirPhone Screener module: Take calls, add notes & names, hold calls, transfer calls, hangup calls. Everything from multiple PCs to multiple SIP clients/phones. Even over openvpn. But as I said: It's not completed by now and has still some features missing. Best regards Sascha Ludwig ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
[RDD] Voip Based Call Manager
Hi all, just wondering if anyone knows of any open source voip based call managers like Phonebox? Have been trying to find a nice solution for this. Cheers Lee ___ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
