The problem is that FS can not locate user So my question is what else I need to modify in order to bridge the call also to extension at the same time? I did lots of Googling, checked the wiki and tried various ways but still without success.

Any advices please? Many Thanks! Hi Michael, Thanks for your suggestion! I tried your suggestion. I saw the softphone extension immediately hung up after it was reached, the I heard "bye" and the extension hung up. Search everywhere only in this topic. Advanced Search. Classic List Threaded. How to bridge a call to an extension defined in dialplan. Liu, Xingkun. Re: How to bridge a call to an extension defined in dialplan.

The softphone is terminated immediately after it is reached. Now I tried it at a different machine with different version of FS. The behaviour is slightly different. This time the softphone is ringing but the recording extension is not correctly executed. A bit desperately to solve this problem ASAP as we will have a demo in a few days. Please could anybody give any clue? Many thanks! Peter Olsson. Can you explain what you're trying to achieve by doing this?

If you bridge to multiple destinations, the first destination that answers will create the actual call - all other legs are terminated. Hi Peter, Thanks for your reply! You are right, that is what I have been experiencing but not I wanted.

Actually my goal is simply to record the user call by my app js or java via ESL and at the same time I want to hear what the user is saying through my computer's speaker.

What I am doing now is to bridge the call to my app for recording and to a softphone for hearing user's voice.

FreeSWITCH with Fred - CDR processing Part. 1

I thought it would work as the wiki says the bridge can bridge the call to multiple endpoints at the same time, but unfortunately it won't work for me. Any suggestions to achieve my goal please: recording and hearing user's voice?

In reply to this post by Liu, Xingkun. Hector Geraldino In reply to this post by mercutioviz. RE: [Freeswitch-users] How to bridge a call to an extension defined in dialplan Hi Michael, Thanks for your suggestion!

Okay, go ahead and pastebin your console debug log of the call from start to finish and also the relevant dialplan code. Use pastebin. Oh yeah, hehe, now I get it. You can't bypass media in a loopback. I probably should have asked these questions before: Why are you trying to bypass media at all?By using our site, you acknowledge that you have read and understand our Cookie PolicyPrivacy Policyand our Terms of Service.

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What is the meaning of internal profile or external profile in Free Switch? It will be really helpful if someone could give me a proper explanation, or at least, point me to a right direction. Those are usually defined in the default configuration of freeswitch. They are sample configurations optimized for internal or external access, you can define other sip profiles with a configuration according to your needs.

Learn more. Asked 8 years, 5 months ago.

freeswitch bridge application

Active 7 years, 6 months ago. Viewed 4k times. Active Oldest Votes. Application bridge connects two channels end-points together. Lallo Lallo 71 1 1 bronze badge.

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Technical site integration observational experiment live on Stack Overflow. Dark Mode Beta - help us root out low-contrast and un-converted bits. Question Close Updates: Phase 1. Related 0. Hot Network Questions. Question feed. Stack Overflow works best with JavaScript enabled.Bridge a new channel to the existing one.

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. Multiple endpoints simultaneously call blast -- no limit to concurrency, first one to answer wins. Bridge the incoming call to extension and Separate multiple endpoints with a comma.

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:. 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:. See full discussion at Bypass Media Overview.

No media mode is an SDP Passthrough feature that permits two endpoints that can see each other no funky N. When set, the media RTP from the originating endpoint is sent directly to the destination endpoint and vice versa. The following example sets them before executing the bridge action. See more about caller ID privacy options. 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. By using commas to separate the addresses, bridge will dial them simultaneously.

Using pipes, it'll dial one at a time. If you need to set different channel variables for each destination, you may prefix the destinations with [] and the variables inside the brackets.

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. These special channels are specified where a normal endpoint would be specified, such as sofia, but provide customized treatment of a call.

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.

Loopback is evil and should only be used as a last resort, when no other approach is possible. 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. Be aware that the user channel should not be used along with forked dialing.By using our site, you acknowledge that you have read and understand our Cookie PolicyPrivacy Policyand our Terms of Service. The dark mode beta is finally here. Change your preferences any time.

Stack Overflow for Teams is a private, secure spot for you and your coworkers to find and share information. The channel seems to go into a hibernate state, but I have housekeeping to do after the bridge is finished which I simply cannot do. Same happens if i do the bridge in XML dialplan, immediately carries on causing my housekeeping to fire early.

I have a workaround, but would still like an answer to the question if anyone has one ie how to stop the bridge exiting immediately. My problem with this is the uncertainty of playing with variables from a session that appears to have gone away from a script that fires at some point in the future.

Learn more. Asked 1 year, 8 months ago. Active 1 year, 8 months ago. Viewed times. This snippet - freeswitch. David Wylie. David Wylie David Wylie 5 5 silver badges 21 21 bronze badges.

Active Oldest Votes. In housekeeping. Anyway, it'll have to do until I can find out how to keep control inside the original script. Did you ever find a better solution? I'm strugging with the same issue.

For me it actually works just fine so I stopped looking for an alternative.

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You could probably session:sleep in a while loop that checks for some signal that the call has gone, but that's speculation and I've not tried it. Sign up or log in Sign up using Google. Sign up using Facebook. Sign up using Email and Password. Post as a guest Name. Email Required, but never shown.

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freeswitch bridge application

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Easiest ways to get up and running. The Pre-built virtual machine has both a written and video how-to to help you get started. All you need is a copy of virtualbox and you can be off and running in minutes. Learn More. If you already have a spare Debian instance handy, you can follow the link below to get detailed instructions on how to get our packages installed.

If you are an experienced developer you can also follow the instructions below to compile the FreeSWITCH platform from source among other manual installation options. Will Apps save us during future pandemics? Could smartphones track the spread of coronavirus? Contact tracing - [Search everywhere only in this topic.

Advanced Search. Classic List Threaded. Jerry Richards. Difference Between "bridge" and "transfer". What is the difference between "bridge" and "transfer"? I'm looking at the demo IVRs. Re: Difference Between "bridge" and "transfer". Brian West. Andrew Thompson In reply to this post by mercutioviz. That depends Is the other leg an inbound call to a toll-free number?

How to bridge a call to an extension defined in dialplan

Will I then be charged for the b-leg? Depends dos your provider charge you for outbound calls? Most do In reply to this post by Aza1. You are making the call on the b-leg. If you are normally charged for making a call, then you will be charged. A call is a call. It doesn't matter if your cousin is making the call, your grandmother is making the call or FS is making the call. It is all happening on your PSTN line.

The telephone company doesn't care who is making the call. They only care that your line is being used to make a call. Ognjen Seslija. This happens when most of my customers forward call from their sip phones to pstn extensions mobile phones i.

FS will then bridge two pstn calls and hence you'll be charged for the b-leg call. Also, most likely is that you will not be able to present the original caller id of a-leg to the forwarded extension, because of pstn settings inbound acls for clid on your line. There is a way on most ISDN lines to "deflect" call back to the telco switch with the forward extension information, so the switch can call the b-leg itself and properly present the original clid.

FS then doesn't need to anything but to respond with the information. I'm pretty sure, though, that telco will charge you for that second call, also. Meftah Tayeb. Billing IVR. In reply to this post by Ognjen Seslija.

Thanks Thanks Ognjen for that extra bit of info. Search everywhere only in this topic Advanced Search Difference Between "bridge" and "transfer". In reply to this post by Aza1 This happens when most of my customers forward call from their sip phones to pstn extensions mobile phones i. Free forum by Nabble.Its core library, libfreeswitch, can be embedded into other projects.

Some dependencies are: [19]. Not all of these software dependencies are required to build the core freeswitch application, but are dependencies of various external modules, such as codecs. FreeSWITCH is a modular application, in which modules can extend the functionality, but the abstraction layer prevents inter-module dependency. The goal is to ensure that one module is not required to load another. Since the raw compressed data is passed through between callers without any processing, this allows support for some codecs that cannot be provided free of charge due to patent or other licensing issues.

freeswitch bridge application

The software supports hardware transcoding cards, such as produced by Sangoma. Some of these codecs are fully licensed, providing an alternative to the pass-through options above. FreeSWITCH provides building blocks from which applications — such as a PBX, a voicemail system, a conferencing system or a calling card — can be built using any of the supported languages.

Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software as open source.

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