- Loopback 2 0 0 – Route Audio Between Applications Pdf
- Loopback 2 0 0 – Route Audio Between Applications Free
- Loopback 2 0 0 – Route Audio Between Applications Near Me
0. About
Bridge a new channel to the existing one (Actually bridge the audio between two sessions). Generally used to route an incoming call to one or more endpoints. Multiple endpoints can be dialed simultaneously or sequentially using the comma and pipe delimiters, respectively.
mod_dptools: bridge VS uuid_bridge
Important: If you're new to LoopBack, use the current release, LoopBack 4.0. To ask questions and discuss how you are using LoopBack, check out the LoopBack Developer Forum. Note: IBM API Connect is an end-to-end API management solution that uses LoopBack to create APIs, and provides integrated build and deployment tools. . TRIAL VERSION: LIMITED TO 5 USES (MESSAGES) PER SESSION. Hear GPS directions or text message notifications while listening to radio/CD in your car. Uses SCO/HFP profile to route audio as phone call and automatically cut your music. AudioBT is an application that unleashes the power of your in-car handsfree Bluetooth (BT) system or mono earpiece. It allows you to hear audio messages. Setting up Loopback. Download Loopback - open the zip and add it your Mac's Applications folder. When you open Loopback, the home menu appears. Select the + New Virtual Device button - 'Loopback Audio' appears (you can change it to another name later on if you want). Under Audio Sources click on the + button, then choose the applications or devices you want to capture audio from. Loopback is a handy macOS app that can be used for a variety of purposes and allows you to create virtual audio devices for merging multiple sound sources and transferring audio between applications. While it may seem a bit complicated at first, things become remarkably simple once you understand exactly.
(uuid_bridge is documented in mod_commands.)
1. Usage
2. Examples
Endpoint examples expressed as API commands:
Simple endpoint:
Multiple endpoints simultaneously (call blast) -- no limit to concurrency, first one to answer wins
Multiple endpoints sequential -- no limit to failover number
Setting Variables for the B-leg simple example using brackets:
Setting Variables for the B-leg thread-global using braces:
Setting Variables for the B-leg with enterprise originate for multiple threads:
2.1 Dialplan Examples
Bridge an incoming call to an external SIP address or termination provider.
Bridge the incoming call to extension 100 and 101. The '%' is used instead of the @ to indicate that the endpoints are registered locally. Separate multiple endpoints with a comma. The ${sip_profile} variable is defined in freeswitch.xml.
To dial multiple contacts all at once:
To dial multiple contacts one at a time:
To bridge an incoming call to an external SIP address or termination provider on the fly which requires authentication without having to create gateways entries:
Extra parameters: for instance, if you are dipping to get an LRN lookup:
or
or Solis 1 0 4 – codes editors integrator edition.
will result in:
However, you will also get 12135551212;rn=12135550000;npdi=yes in the To: parameter. Thus, to get rid of that you can append the carat (^) as follows:
3. Options
You can set different options to modify the behavior of the call. Be sure to set the option before executing the application. Here are some examples:
3.1 Timeout
The maximum number of seconds to wait for an answer state from a remote endpoint.
3.2 No Media Mode
See full discussion at Bypass Media Overview.
Best hard drive format for windows 10. No media mode is an SDP Passthrough feature that permits two endpoints that can see each other (no funky N.A.T.) to connect their media sessions directly while FreeSWITCH maintains control of the SIP signaling. This is useful if you have two end-points that need to use a codec that is currently not supported in FreeSWITCH (video) or if you are using FreeSWITCH in a high performance walled garden network and want to minimize the RTP handling FreeSWITCH is doing to maximize call traffic.
When set, the media (RTP) from the originating endpoint is sent directly to the destination endpoint and vice versa. The signaling (SIP) for both endpoints still goes through FreeSWITCH, but the media is point-to-point.
Before executing the bridge action you must set the 'bypass_media' flag to true.
bypass_media must only be set on the A leg of a call, for example:
3.3.Setting Outgoing CallerID
If you are using FreeSWITCH as a PBX you may want to control the outgoing CallerID that is sent to the PSTN or your SIP provider. Your SIP Provider will most likely require you to use a specific CallerID number (or a userid instead). The following example sets them before executing the bridge action. See more about caller ID privacy options.
3.4 Sending Ringback
You may want to simulate ringback to your internal users while you dial a provider, or you may need to force a ringback back upstream when you are dialing multiple extensions and cannot determine what call treatment you will need to provide yet. In order to accomplish this, you need to set a chanvar before going to the bridge application.
Note that you can use another ring than ${us-ring}, but it would have to be defined in your configuration (typically in vars.xml).
3.5 Calling multiple destinations
By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialled in a multi-threaded manner (this is referred to as 'Enterprise Origination') - this gives more flexibility (and avoids the 'Only calling the first element in the list in this mode' warning)
If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example:
Note: by default when bridging, the first endpoint to provide media (as opposed to actually answering) will win, and the other endpoints will stop ringing. For internal endpoints, this usually doesn't matter. However, in the case of cell phone providers, any custom music that plays for the caller while ringing counts as media. In some cases, the ringing sound itself is media. If your bridge command includes a cell phone number and your internal endpoints stop ringing as soon as the cell phone starts, you will need to enable the 'ignore_early_media' option:
Loopback 2 0 0 – Route Audio Between Applications Pdf
3.6 Implementing Failover
Failover for your outbound gateway is easy to implement at bridge time using the | separator:
Setting the options ping parameter on the gateway in the sip_profiles definition will allow FreeSWITCH to determine a gateway has failed ahead of time which allows the bridge to go to the secondary immediately rather than waiting for a timeout during call setup.
3.7 Call Camping
See Call Camping.
4. Special channels
These special channels are specified where a normal endpoint would be specified, such as sofia, but provide customized treatment of a call.
4.1 error
You can bridge the call to the error channel in order to specify a hangup cause.
4.2 gateway
When dialing using the gateway channel, it will use the authentication details specified in the sip_profiles definition if challenged.
4.3 group
The group special channel will dynamically create a dial string to reach all endpoints defined as part of a group in the directory by using the 'group_call' function.
This is the same as:
4.4 loopback
Loopback creates a pseudo-endpoint that starts a new pass through the specified dialplan, but can cause unusable CDR entries as a result. Do not use unless there is no other option, but expect to spend time troubleshooting it.
This example searches the inherited dialplan for extension 1000 in the inherited context.
Loopback is evil and should only be used as a last resort, when no other approach is possible. - anthm
For more info look at Loopback endpoint
4.5 user
[ Note: This section was imported directly from the old wiki and must be edited to the rest of the text under 'user' to remove redundancies. ]
If the dial-string param is set for a user or a domain you can bridge a call to the user endpoint without going through a profile. The default dial-string param in the example config files for FreeSWITCH has the following syntax:
When this param is set on the destination user you can bridge to it this way:
When you use the user channel the user is pulled from your domain in your directory then the main domain is searched for a tag. If that is not found it searches in the for a tag. If a tag is located, it looks for the param 'dial-string' as the originate string to use for that user. The special variables ${dialed_user} and ${dialed_domain} are available in this string for expansion based on the user and domain specified.
for instance if all calls are sip you could set the dial-string in your whole domain as
Be aware that the user channel should not be used along with forked dialing. Instead use the following in your data= for the bridge:
Since FreeSWITCH has a user directory, you can save how to reach every user in the user's directory entry by saving it in its 'dial-string' parameter. When dialing using the user channel, it will lookup the dial-string stored in the extension's directory entry and dial it instead.
If the user has no dial-string, the default dial-string specified in the directory entry for the channel's domain is used instead.
If you want to route to many user channels, you will have to separate them by :_: as opposed to the comma ',' to generate an originate command for each user.
Note again: You have to be careful with whitespace here generally. For some reason, FS will see user [bob@domain ] and NOT [bob@domain] which will make it not find the user, if you try to be fancy like for example this: Picture instruments chroma mask 2 0 101.
So be careful about formatting.
5. See Also
- Loopback endpoint — a special user channel for very specialized purposes
- Channel variables — dicsusses how channel variables, especially loopback_bowout, can be set and how they affect this command
Loopback 2 0 0 – Route Audio Between Applications Free
To Computer / From Computer Audio Streams
MOTU AVB audio interfaces provide powerful mixing and routing options to meet a variety of audio scenarios. The Routing matrix found in the Routing Tab of the MOTU AVB Discovery App allows you to route signal to and from all of the available inputs and outputs on your interface.
mod_dptools: bridge VS uuid_bridge
Important: If you're new to LoopBack, use the current release, LoopBack 4.0. To ask questions and discuss how you are using LoopBack, check out the LoopBack Developer Forum. Note: IBM API Connect is an end-to-end API management solution that uses LoopBack to create APIs, and provides integrated build and deployment tools. . TRIAL VERSION: LIMITED TO 5 USES (MESSAGES) PER SESSION. Hear GPS directions or text message notifications while listening to radio/CD in your car. Uses SCO/HFP profile to route audio as phone call and automatically cut your music. AudioBT is an application that unleashes the power of your in-car handsfree Bluetooth (BT) system or mono earpiece. It allows you to hear audio messages. Setting up Loopback. Download Loopback - open the zip and add it your Mac's Applications folder. When you open Loopback, the home menu appears. Select the + New Virtual Device button - 'Loopback Audio' appears (you can change it to another name later on if you want). Under Audio Sources click on the + button, then choose the applications or devices you want to capture audio from. Loopback is a handy macOS app that can be used for a variety of purposes and allows you to create virtual audio devices for merging multiple sound sources and transferring audio between applications. While it may seem a bit complicated at first, things become remarkably simple once you understand exactly.
(uuid_bridge is documented in mod_commands.)
1. Usage
2. Examples
Endpoint examples expressed as API commands:
Simple endpoint:
Multiple endpoints simultaneously (call blast) -- no limit to concurrency, first one to answer wins
Multiple endpoints sequential -- no limit to failover number
Setting Variables for the B-leg simple example using brackets:
Setting Variables for the B-leg thread-global using braces:
Setting Variables for the B-leg with enterprise originate for multiple threads:
2.1 Dialplan Examples
Bridge an incoming call to an external SIP address or termination provider.
Bridge the incoming call to extension 100 and 101. The '%' is used instead of the @ to indicate that the endpoints are registered locally. Separate multiple endpoints with a comma. The ${sip_profile} variable is defined in freeswitch.xml.
To dial multiple contacts all at once:
To dial multiple contacts one at a time:
To bridge an incoming call to an external SIP address or termination provider on the fly which requires authentication without having to create gateways entries:
Extra parameters: for instance, if you are dipping to get an LRN lookup:
or
or Solis 1 0 4 – codes editors integrator edition.
will result in:
However, you will also get 12135551212;rn=12135550000;npdi=yes in the To: parameter. Thus, to get rid of that you can append the carat (^) as follows:
3. Options
You can set different options to modify the behavior of the call. Be sure to set the option before executing the application. Here are some examples:
3.1 Timeout
The maximum number of seconds to wait for an answer state from a remote endpoint.
3.2 No Media Mode
See full discussion at Bypass Media Overview.
Best hard drive format for windows 10. No media mode is an SDP Passthrough feature that permits two endpoints that can see each other (no funky N.A.T.) to connect their media sessions directly while FreeSWITCH maintains control of the SIP signaling. This is useful if you have two end-points that need to use a codec that is currently not supported in FreeSWITCH (video) or if you are using FreeSWITCH in a high performance walled garden network and want to minimize the RTP handling FreeSWITCH is doing to maximize call traffic.
When set, the media (RTP) from the originating endpoint is sent directly to the destination endpoint and vice versa. The signaling (SIP) for both endpoints still goes through FreeSWITCH, but the media is point-to-point.
Before executing the bridge action you must set the 'bypass_media' flag to true.
bypass_media must only be set on the A leg of a call, for example:
3.3.Setting Outgoing CallerID
If you are using FreeSWITCH as a PBX you may want to control the outgoing CallerID that is sent to the PSTN or your SIP provider. Your SIP Provider will most likely require you to use a specific CallerID number (or a userid instead). The following example sets them before executing the bridge action. See more about caller ID privacy options.
3.4 Sending Ringback
You may want to simulate ringback to your internal users while you dial a provider, or you may need to force a ringback back upstream when you are dialing multiple extensions and cannot determine what call treatment you will need to provide yet. In order to accomplish this, you need to set a chanvar before going to the bridge application.
Note that you can use another ring than ${us-ring}, but it would have to be defined in your configuration (typically in vars.xml).
3.5 Calling multiple destinations
By using commas to separate the addresses, bridge will dial them simultaneously. Using pipes, it'll dial one at a time. Use :_: to separate multiple destinations to be dialled in a multi-threaded manner (this is referred to as 'Enterprise Origination') - this gives more flexibility (and avoids the 'Only calling the first element in the list in this mode' warning)
If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets. Example:
Note: by default when bridging, the first endpoint to provide media (as opposed to actually answering) will win, and the other endpoints will stop ringing. For internal endpoints, this usually doesn't matter. However, in the case of cell phone providers, any custom music that plays for the caller while ringing counts as media. In some cases, the ringing sound itself is media. If your bridge command includes a cell phone number and your internal endpoints stop ringing as soon as the cell phone starts, you will need to enable the 'ignore_early_media' option:
Loopback 2 0 0 – Route Audio Between Applications Pdf
3.6 Implementing Failover
Failover for your outbound gateway is easy to implement at bridge time using the | separator:
Setting the options ping parameter on the gateway in the sip_profiles definition will allow FreeSWITCH to determine a gateway has failed ahead of time which allows the bridge to go to the secondary immediately rather than waiting for a timeout during call setup.
3.7 Call Camping
See Call Camping.
4. Special channels
These special channels are specified where a normal endpoint would be specified, such as sofia, but provide customized treatment of a call.
4.1 error
You can bridge the call to the error channel in order to specify a hangup cause.
4.2 gateway
When dialing using the gateway channel, it will use the authentication details specified in the sip_profiles definition if challenged.
4.3 group
The group special channel will dynamically create a dial string to reach all endpoints defined as part of a group in the directory by using the 'group_call' function.
This is the same as:
4.4 loopback
Loopback creates a pseudo-endpoint that starts a new pass through the specified dialplan, but can cause unusable CDR entries as a result. Do not use unless there is no other option, but expect to spend time troubleshooting it.
This example searches the inherited dialplan for extension 1000 in the inherited context.
Loopback is evil and should only be used as a last resort, when no other approach is possible. - anthm
For more info look at Loopback endpoint
4.5 user
[ Note: This section was imported directly from the old wiki and must be edited to the rest of the text under 'user' to remove redundancies. ]
If the dial-string param is set for a user or a domain you can bridge a call to the user endpoint without going through a profile. The default dial-string param in the example config files for FreeSWITCH has the following syntax:
When this param is set on the destination user you can bridge to it this way:
When you use the user channel the user is pulled from your domain in your directory then the main domain is searched for a tag. If that is not found it searches in the for a tag. If a tag is located, it looks for the param 'dial-string' as the originate string to use for that user. The special variables ${dialed_user} and ${dialed_domain} are available in this string for expansion based on the user and domain specified.
for instance if all calls are sip you could set the dial-string in your whole domain as
Be aware that the user channel should not be used along with forked dialing. Instead use the following in your data= for the bridge:
Since FreeSWITCH has a user directory, you can save how to reach every user in the user's directory entry by saving it in its 'dial-string' parameter. When dialing using the user channel, it will lookup the dial-string stored in the extension's directory entry and dial it instead.
If the user has no dial-string, the default dial-string specified in the directory entry for the channel's domain is used instead.
If you want to route to many user channels, you will have to separate them by :_: as opposed to the comma ',' to generate an originate command for each user.
Note again: You have to be careful with whitespace here generally. For some reason, FS will see user [bob@domain ] and NOT [bob@domain] which will make it not find the user, if you try to be fancy like for example this: Picture instruments chroma mask 2 0 101.
So be careful about formatting.
5. See Also
- Loopback endpoint — a special user channel for very specialized purposes
- Channel variables — dicsusses how channel variables, especially loopback_bowout, can be set and how they affect this command
Loopback 2 0 0 – Route Audio Between Applications Free
To Computer / From Computer Audio Streams
MOTU AVB audio interfaces provide powerful mixing and routing options to meet a variety of audio scenarios. The Routing matrix found in the Routing Tab of the MOTU AVB Discovery App allows you to route signal to and from all of the available inputs and outputs on your interface.
Along with the physical inputs and outputs on the device itself, the built-in 48-channel on-board digital mixer, and any enabled banks of AVB audio streams, the To Computer and From Computer audio streams are important I/O options to be familiar with, as they are required for sending signal to or from your computer.
The Audio Interface device preset provides a default setting to send audio to and from your computer, however for further control, a deeper understanding of the relationship between these audio streams and your computer is necessary.
Before doing any routing, if you intend to send audio to a recording application (like Digital Performer or any other DAW), OR intend to play audio back from your computer in any way (even through applications as simple as iTunes, Quicktime, or when streaming content from the internet), you need to enable To Computer and/or From Computer audio streams. In the Device Tab of the MOTU AVB Web App, scroll down to the Computer Setup section, and set the desired number of To Computer and From Computer audio streams.
Here are a few considerations for deciding how many audio streams are necessary:
- If you are simply playing audio through any application on your computer, other than your DAW, you likely only need two From Computer streams enabled for stereo playback.
- A larger number of From Computer audio streams are necessary when working in your DAW, and you intend for the ability to route tracks in your project to multiple destinations, (more than one stereo stream).
- Enabled To Computer audio streams should at least equal the number of individual audio tracks you would like to record at once in your DAW, (stereo tracks equal 2 streams).
- Generally, there is no issue with enabling more streams than necessary, however this could contribute to high DSP usage if other features on your AVB interface are simultaneously in use.
Via USB Connection*Via ThunderBolt Connection
Note: As of the firmware version v1.2.0, all AVB models have three USB Modes. You can enable 24 channels (in and out) at sample rates 44.1 - 192 kHz, 32 channel at sample rates 44.1 - 96 kHz, or 64 channels at sample rates 44.1 - 48 kHz. You can define which USB Mode the interface is running in from the Configuration Section of the Device Tab. Be sure to download the latest firmware for your unit, which you can find here.
When connecting your AVB interface via ThunderBolt, you can access to up to 128 channels in and out.
Once audio streams are enabled in the Computer Setup section, swing over to the Routing Tab of the MOTU AVB Discovery App. You will now find a 'From Computer' bank of inputs listed across the top of the routing grid, and a 'To Computer' bank of outputs listed on the left; (USB) or (TB) will be added to its title to specify the connection method being used between your interface and the computer.
In most examples, you will generally only use the 'From Computer 1-2' audio streams. This stereo pair will carry the audio coming from your computer (if your AVB interface is set as the primary device for system's sound playback), or if you have designated the AVB interface as the device for your DAW's session.
Expand the From Computer input bank to see each From Computer audio stream, and make a connection between the From Computer 1-2 audio streams, and the physical output channels you want to monitor playback through. (It is generally easiest to expand the output bank on the left side as well, so you can see the individual output channels.) For example, click on the intersection between From Computer 1 and Main Out L, as well as the intersection between From Computer 2 and Main Out R to route your computer's playback to the Main Outs, as shown below.
Once done, when playing audio through iTunes for example, you will see signal activity on the From Computer 1-2 indicator lights and matching signal on the Main Out 1-2's indicator lights.
Often, you may want the same stereo signal to be sent to multiple outputs at the same time. To quickly achieve this, patch the same From Computer 1-2 audio streams to another pair of physical outputs such as the two sets of Phones and the Monitor outputs on the 1248. This stereo signal can also be sent to the Mix In channels (not shown) to feed the interface's on-board mixer for further processing.
If you are working with a larger number of From Computer audio streams (perhaps to route multiple tracks in your DAW to different physical outputs on your interface), the same theory applies. Each From Computer audio stream can be routed directly to its own physical output.
For example, if you own a 1248, and want to process audio from your DAW through a hardware signal processor, each track in your DAW can be routed to its own From Computer audio stream. Then in the MOTU AVB Web App, route each From Computer audio stream directly to one of the 1248's Analog Outputs as shown below.
Configuring the 'To Computer' audio streams is done in a similar manner and is required for your computer to 'see' any input from your device.
Loopback 2 0 0 – Route Audio Between Applications Near Me
MOTU AVB Theory: When you plug a source into a physical input on your interface, this input is not automatically detected by your DAW or other computer application. This is because your AVB interface can take that input signal and send it anywhere else (on-board mixer, AVB streams, etc.) without the computer ever accessing it. The computer is a specific destination where signal can be sent to, and sending signal to the computer needs to be manually configured. This is why 'To Computer' audio streams are considered outputs.
Expand the 'To Computer' bank on the left, and the bank of physical inputs on your unit you want to record. Click on the intersection between Analog 1 and To Computer 1, Analog 2 and To Computer 2, etc. Now each input is being routed to its own virtual 'To Computer' audio stream, which you can assign as the input to an audio track in your DAW.
Keep in mind, the same input, (Analog 1, for example), can be sent to multiple 'To Computer' audio streams for simultaneous recording on multiple tracks. However, each 'To Computer' audio stream can only receive signal from one input.
Tripura rahasya telugu pdf. To explain this visually, this means you can make vertical lines in the Routing grid when routing inputs to the 'To Computer' streams, but you cannot make horizontal lines.
To check your signal flow when a connection is made, any signal identified by your input channel's indicator lights will also be visible on the 'To Computer' stream's light on the left.
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