ConfBridge ConfigurationConfBridge Profiles and Menus are configured in the confbridge.conf configuration file - normally located at /etc/asterisk/confbridge.conf. The file contains three reserved sections:. general. defaultbridge. defaultuserThe general section is currently unused, but is reserved for future use.The defaultbridge section contains all options invoked when ConfBridge is instantiated from the dialplan without a bridge profile argument.The defaultuser section contains all options invoked when ConfBridge is instantiated from the dialplan without a user profile argument.Each section contains a type definition. The type definition determines the function of the section. The three types are:.
bridge. user. menubridge is used to denote Bridge Profile section definitions.user is used to denote User Profile section definitions.menu is used to denote Conference Menu section definitions.All other sections, defined by a section identifier encapsulated in square brackets, are user-definable.
OptionValuesDescriptionNotestypebridgeSet this to bridge to configure a bridge profilemaxmembersinteger; e.g. 50Limits the number of participants for a single conference to a specific number. By default, conferences have no participant limit. After the limit is reached, the conference will be locked until someone leaves.
Admin-level users are exempt from this limit and will still be able to join otherwise-locked, because of limit, conferences.recordconferenceyes/noRecords the conference call starting when the first user enters the room, and ending when the last user exits the room. The default recorded filename is 'confbridge-.wav and the default format is 8kHz signed linear.
By default, this option is disabled. This file will be located in the configured monitoring directory as set in asterisk.confrecordfilepath, e.g. /tmp/myfilesWhen recordconference is set to yes, the specific name of the recorded file can be set using this option. Note that since multiple conferences may use the same Bridge profile, this can cause issues, depending on the configuration. It is recommended to only use this option dynamically with the CONFBRIDGE dialplan function.
This allows the recorded name to be specified and a unique name to be chosen. By default, the recorded file is stored in Asterisk's spool/monitory directory, with a unique filename starting with the 'confbridge' prefix.internalsamplerateauto, 8000, 12000, 16000, 24000, 32000, 44100, 48000, 96000, 192000Sets the internal native sample rate at which to mix the conference. The 'auto' option allows Asterisk to adjust the sample rate to the best quality / performance based on the participant makeup.
Numbered values lock the rate to the specified numerical rate. If a defined number does not match an internal sampling rate supported by Asterisk, the nearest sampling rate will be used instead.mixinginterval10, 20, 40, 80Sets, in milliseconds, the internal mixing interval. By default, the mixing interval of a bridge is 20ms. This setting reflects how 'tight' or 'loose' the mixing will be for the conference. Lower intervals provide a 'tighter' sound with less delay in the bridge and consume more system resources. Higher intervals provide a 'looser' sound with more delay in the bridge and consume less resourcesvideomodenone, followtalker, lastmarked, firstmarkedConfigured video (as opposed to audio) distribution method for conference participants. Participants must use the same video codec.
Confbridge does not provide MCU functionality. It does not transcode, scale, transrate, or otherwise manipulate the video. OptionValuesDescriptionNotestypeuserSet this to user to configure a user profileadminyes/noSets if the user is an Admin or not. By default, no.markedyes/noSets if the user is Marked or not. By default, no.startmutedyes/nosets if the user should start out muted. By default, no.musiconholdwhenemptyyes/noSets whether music on hold should be played when only one person is in the conference or when the user is waiting on a marked user to enter the conference. By default, off.musiconholdclassmusic on hold classSets the music on hold class to use for music on hold.quietyes/noWhen set to 'yes,' enter/leave prompts and user introductions are not played.
By default, no.announceusercountyes/noSets if the number of users in the conference should be announced to the caller. By default, no.announceusercountallyes/no; or an integerSets if the number of users should be announced to all other users in the conference when someone joins. When set to a number, the announcement will only occur once the user count is above the specified numberannounceonlyuseryes/noSets if the only user announcement should be played when someone enters an empty conference. By default, yes.announcementfilenameIf set, the sound file specified by filename will be played to the user, and only the user, upon joining the conference bridge.New in Asterisk 11waitmarkedyes/noSets if the user must wait for another marked user to enter before joining the conference.
By default, no.endmarkedyes/noIf enabled, every user with this option in their profile will be removed from the conference when the last marked user exists the conference.dspdropsilenceyes/noDrops what Asterisk detects as silence from entering into the bridge. Enabling this option will drastically improve performance and help remove the buildup of background noise from the conference. This option is highly recommended for large conferences, due to its performance improvements.dsptalkingthresholdinteger in millisecondsThe time, in milliseconds, by default 160, of sound above what the DSP has established as base-line silence for a user, before that user is considered to be talking. This value affects several options:. Sinvr full game cracked download. Audio is only mixed out of a user's incoming audio stream if talking is detected. If this value is set too loose, the user will hear themselves briefly each time they begin talking until the DSP has time to establish that they are in fact talking.
When talker detection AMI events are enabled, this value determines when talking has begun, which causes AMI events to fire. If this value is set too tight, AMI events may be falsely triggered by variants in the background noise of the caller. The dropsilence option depends on this value to determine when the user's audio should be mixed into the bridge after periods of silence.
If this value is too loose, the beginning of a user's speech will get cut off as they transition from silence to talking.dspsilencethresholdinteger in millisecondsThe time, in milliseconds, by default 2500, of sound falling within what the DSP has established as the baseline silence, before a user is considered to be silent. The best way to approach this option is to set it slightly above the maximum amount of milliseconds of silence a user may generate during natural speech.
This value affects several operations:. When talker detection AMI events are enabled, this value determines when the user has stopped talking after a period of talking. If this value is set too low, AMI events indicating that the user has stopped talking may get faslely sent out when the user briefly pauses during mid sentence. The dropsilence option depends on this value to determine when the user's audio should begin to be dropped from the bridge, after the user stops talking. If this value is set too low, the user's audio stream may sound choppy to other participants.talkdetectioneventsyes/noSets whether or not notifications of when a user begins and ends talking should be sent out as events over AMI.
By default, no.denoiseyes/noWhether or not a noise reduction filter should be applied to the audio before mixing. By default, off. This requires codecspeex to be built and installed. Do not confuse this option with dropsilence.
Denoise is useful if there is a lot of background noise for a user, as it attempts to remove the noise while still preserving the speech. This option does not remove silence from being mixed into the conference and does come at the cost of a slight performance hit.jitterbufferyes/noWhether or not to place a jitter buffer on the caller's audio stream before any audio mixing is performed. This option is highly recommended, but will add a slight delay to the audio and will incur a slight performance penalty. This option makes use of the JITTERBUFFER dialplan function's default adaptive jitter buffer.
For a more fine-tuned jitter buffer, disable this option and use the JITTERBUFFER dialplan function on the calling channel, before it enters the ConfBridge application.pinintegerSets if the user must enter a PIN before joining the conference. The user will be prompted for the PIN.announcejoinleaveyes/noWhen enabled, this option prompts the user for their name when entering the conference. After the name is recorded, it will be played as the user enters and exists the conference.
By default, no.dtmfpassthroughyes/noWhether or not DTMF received from users should pass through the conference to other users. By default, no.ExampleIn this example, we will create a user profile called 'fancyuser' that includes music on hold, user count announcements, join/leave announcements, silence detection, noise reduction and requires a PIN of 456. OptionValuesDescriptionNotestypemenuSet this to menu to configure a conference menuplayback(&&.)Plays back an audio file, or a string of audio files chained together using the & character, to the user and then immediately returns them to the conference.playbackandcontinue(&&.)Plays back an audio file, or a series of audio files chained together using the & character, to the user while continuing the collect the DTMF sequence. This is useful when using a menu prompt that describes all of the menu options. Note that any DTMF during this action will terminate the prompt's playback.togglemuteToggles mute on and off. When a user is muted, they will not be able to speak to other conference users, but they can still listen to other users. While muted, DTMF keys from the caller will continue to be collected.noopThis action does nothing.
Its only real purpose exists for being able to reserve a sequence in the configuration as a menu exit sequence.decreaselisteningvolumeDecreases the caller's listening volume. Dell km636 wireless keyboard manual. Kami nomi zo shiru sekai manga indo. Everything they hear will sound quieter.increaselisteningvolumeIncreases the caller's listening volume. Everything they hear will sound louder.resetlisteningvolumeResets the caller's listening volume to the default level.decreasetalkingvolumeDecreases the caller's talking volume. Everything they say will sound quieter to other callers.increasetalkingvolumeIncreases the caller's talking volume.
Everything they say will sound louder to other callers.resettalkingvolumeResets the caller's talking volume to the default level.dialplanexec(context,exten,priority)Allows one to escape from the conference and execute commands in the dialplan. Once the dialplan exits, the user will be put back into the conference.leaveconferenceAllows a user to exit the conference and continue execution in the dialplan.adminkicklastAllows an Admin to remove the last participant from the conference. This action only works for users whose User Profiles set them as conference Admins.admintoggleconferencelockAllows an Admin to toggle locking and unlocking the conference. When the conference is locked, only other Admin users can join. When the conference is unlocked, any user may join up to the limit defined by the maxmembers Bridge Profile option. This action only works for users whose User Profiles set them as conference Admins.admintogglemuteparticipantsAllows an Admin to mute/unmute all non-admin participants in the conference.New in Asterisk 11setassinglevideosrcAllows a user to set themselves as the single video distribution source for all other participants. This overrides the videomode setting.releaseassinglevideosrcAllows a user to release themselves as the single video source.
Upon release of the video source, and/or if videomode is set to 'none,' this action will result in the conference returning to whatever video mode the Bridge Profile is using. This action will have no effect if the user is not currently the video source. The user is also not guaranteed that the use of this action will prevent them from becoming the video source later.participantcountPlays back the current number of participants into the conference.New in Asterisk 11ExampleIn this example, we'll create a menu called 'fancymenu.' This menu will utilize many of the options listed above. We will construct a features menu that plays when the user enters the. character.
Since we will do this using the playbackandcontinue option, we will define other menu items as being a 'subset' of the. command, e.g.4, so that once the user presses., they can listen to the menu options and then press the specific 'after-star' option, e.g.
4, to affect the option. Additionally, we will duplicate those same sub-features as non-. features, so that the user does not need to have entered the. menu structure in order to affect the options, they can just press the key, e.g. '4' at any time, regardless of whether or not they're in the.-tree.
I have largely ignored fax over VoIP for a long time, because it’s not something I do. At PSU, we attach fax machines to analog lines; done. But recently, I was asked to assist with a quick-turnaround fax project: send a fax to 4,059 analog lines on campus, calls spaced one minute apart, and record the results, so that we can see how many fax machines are hooked up out there. I tried module, but couldn’t get it to communicate. Next I built the spandsp module, and told Asterisk to use it, and it worked right away. A Perl script using the excellent Asterisk::AMI module (referenced previously on this blog) drove the dialer, and standard Asterisk CDR with a few extra fields recorded the fax results. By the way, the faxes were sent in audio mode (not T.38) using SIP and G.711u over the Internet, and only 3 out of 4,059 calls failed.
It’s been a too-busy-to-write week, but I found some time the last few evenings to brush up my C coding (only hacking at this point, really) to make Asterisk 1.8 do something I’ve wanted from the beginning: Google Talk’s invisible mode for the Google Voice integration.Quick background: Google wrote a variety of XMPP extensions to make Google Talk do voice and a bunch of other non-standard XMPP things. One of those things is what Google calls “shared statuses,” which also includes invisible mode–the ability to be online but set your client as invisible so that others can’t see you, but could chat with you.You might want to use this feature if, like me, your Google Voice number is attached to your main GMail account. I don’t use the chat features but I appear online to anyone who has me in their contact list, as long as Asterisk is logged in, and that has the potential to draw unwanted instant messages (that will never get answered).Note: I want to reinforce that this is a hack. The code is clean enough, but this is only useful for Google Talk and is only implemented as a set of Asterisk CLI commands. Furthermore, this functionality would probably never be implemented in Asterisk’s resjabber because it’s Google-proprietary. On the other hand, if any Asterisk developers want to run with this, go for it!
I don’t guarantee that this code will work for you and if your Asterisk process segfaults, stop using it.SetupApply against Asterisk 1.8.2.2’s (or trunk, currently) res/resjabber.c by to your Asterisk source directory and running patch information at login rather than Google’s special shared-status presence scheme. If you want to be sure you are invisible, just start off by setting invisible off and then set it on. For me, that works every time. (I’d be grateful for someone who really knows all about this to fix it up and make it work right, the first time.)ReferencesI got the idea from, which refers to the on how it works. Go to the and get the.
Put it in your Asterisk source directory (the root, where the configure script, Makefile, and other such files exist). Call it gtalkfix.patch, to be a little less ambiguous. From that same directory, run the command patch. As, Google Voice seems to have stopped accepting outbound calls from Asterisk 1.8.
There’s a on this issue, and in user forums.I fired up again, without updating any code (still the source from mid-October, 1.0.head (git-dc-10-17 16-19-38 -0400)), and it works perfectly. Apparently, the FreeSWITCH guys are using a different method to make the Google Voice connection, or the Google folks are specifically filtering Asterisk or have made a change that affects Asterisk and to which FreeSWITCH is immune. In this post I’ll share the few notes I have on installing Asterisk 1.8 on a. First, note that I am not promoting this specific cloud provider. (There are several others, such as and.) But I am all about VoIP on the cheap, and the baseline RS CloudServer seems to be as inexpensive as you can get. Moreover, I started using RS CloudServers a little while ago just for some quick Linux testing, and liked them. So when I thought about testing Asterisk 1.8, I set up a new instance there.I’m not a software engineer and can’t effectively analyze Asterisk’s code to say whether it should or should not perform well in a cloud environment, but testing in this environment shows that it works fine, with one caveat: you can’t use the DAHDI kernel module.
On other providers where the kernel headers and sources are available, you can compile DAHDI, but RS Cloud does not provide their Xen-optimized kernel source. The good news is that with Asterisk 1.8 (actually, starting in 1.6.2), you don’t need DAHDI unless you have directly-attached telephony hardware, which you don’t, because your server is in the cloud.To further explain that: previously, you’d need a DAHDI hardware module or the DAHDI dummy driver to provide the timer for the IAX2 protocol and MeetMe.
IAX2 no longer needs it, and while MeetMe does still need at least the dummy timer, there’s a new conference bridge application available that doesn’t need it–appconfbridge. (And if you don’t care about having a conference bridge anyway, this whole matter is moot.)After all that introduction, I’m glad to say that actually building Asterisk and FreePBX on the cloud server is straightforward and only requires a few adjustments from building on physical hardware. I’m going on the assumption that you have built Asterisk and FreePBX before on a physical server. Here are the extra steps I had to take to build on RS Cloud.Asterisk. Create an instance of CentOS 5.5 on the 256 MB RAM/10 GB storage server. For test, this is enough.
If you end up running a lot of calls through it, you’ll need a bigger VM. But the 256 MB server is a bargain at 1.5 cents per hour and 22 cents per gigabyte download and 8 cents per gigabyte upload. (Can you even get old hardware, electricity and bandwidth for testing for this price?). yum update and add build tools to the environment ( yum install gcc make gcc-c ncurses-devel libtool subversion). This is also a good time to add the packages that FreePBX needs.
(My list included mysql-server mysql-devel libxml2-devel libtiff-devel php-pear php-pear-DB php-gd php-mysql php-pdo perl-DateManip modssl). useradd -c 'Asterisk PBX' -d /var/lib/asterisk asterisk; mkdir /var/run/asterisk; mkdir /var/log/asterisk. Get asterisk-1.8.x.tar.gz and expand. There is no more asterisk-addons package; it’s now part of the asterisk tar file.
Freepbx 14 No Conference Room App
For Google Talk/XMPP/Google Voice:. yum install openssl-devel. svn co iksemel-read-only. Build iksemel with./autogen.sh;./configure; make; make install. Iksemel puts its libraries in /usr/local/lib, so add this location to your library path: echo /usr/local/lib /etc/ld.so.conf.d/local.conf; ldconfig.
Build: cd to the asterisk-1.8 directory, then./configure; make menuselect. Review the modules being built. Appmeetme won’t be available because DAHDI isn’t built. Add appconfbridge instead. Make sure resjabber and changtalk are selected if you want to set up Google Talk/Voice. Review the other modules including the ones listed in Add-ons (formerly a separate tar file) especially if you are adding FreePBX–you’ll want cdrmysql. Keep building: make; make install; make samples.
Important for the RS Cloud environment: there is no TTY9, which safeasterisk puts the Asterisk console on. Edit /usr/sbin/safeasterisk and comment out line 5 ( TTY=9) and set CONSOLE=no on line 6. Everything else should be fine. Asterisk should be ready to go!FreePBX. When I installed FreePBX 2.8.0, it complained that Asterisk 1.8 was not supported. The repository has an updated installer (see ) but if you’re going from the 2.8.0 (or earlier?) tarball, you’ll have to make the change highlighted here. Edit line 899 in installamp and change 1.7 to 1.9.
Now install FreePBX according to your own documentation or that available onStartup. Add /usr/local/sbin/amportal start to the end of /etc/rc.d/rc.local, or make an init script for /etc/init.d. I’m lazy and chose the former.
Function Confbridge Not Registered
Start everything up with amportal start.More to doGet familiar with iptables, if you’re not already. This is the way to control access on your RS CloudServer node. By default it permits SSH, so you’ll want to add lines to permit web, SIP, IAX2, and so on.Since you’re running in 256 MB RAM, you might want to pare down the /etc/httpd/conf/httpd.conf and reduce the number of web servers spawned, to save a little memory. The RS CloudServer is already pretty thin, otherwise.TestingI did some additional configuration of FreePBX, not essential to this posting, and then configured a softphone extension and a Google Voice trunk (to be discussed in a later posting). Then I made some calls.
Even with both legs of the call going over the Internet, from a shared hosting facility, using G.711 uLaw (PCMU), the sound was great.LaterMore on the Google Voice setup and FreePBX, and considerations for having your PBX out on the Internet. This would be a good time to start thinking about secure RTP! Posts navigation.
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