Im trying to figure out how blind transfers are supposed to work with ARI. When two channels are bridged together through ARI, and one of them performs a blind SIP transfer, two things happen : – a Local channel is instanciated and goes through the dialp..
Author : Jean Aunis
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Im struggling to find a way to properly handle blind transfers with ARI. This is my use case : – Alice calls Bob through Asterisk – dialing and bridging is done with ARI – when Bob blind-transfers to Charlie, I would like to use the redirect ARI operati..
Ive just gone through the process of cross-compiling Asterisk 16 for ARM. I thought it would be as easy as calling the ./configure script with the appropriate host parameter, but it turned out to be more complicated. Im wondering whether I did someth..
I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled.When I receive a SIP INFO, the logs tell me that a DTMF begin is generated, but no related DTMF end is generated, unless the call is ended. Here is an excerpt..
Has anyone already implemented some sort of call preemption in Asterisk ? I am trying to achieve something like this :- I want to limit the number of calls on a given SIP peer to 10- on the other hand, some calls have higher priority than others- w..
How to play the hold music on a channel in a ConfBridge when all the other channels ar..
I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technol..
I noticed that when a channel is destroyed, two different events can be raised : ChannelDestroyed and ChannelHangupRequest. These two events seem to be mutually exclusive : if I receive a ChannelHangupRequest, I will never receive a ChannelDestroy..
This means the remote end was not sending any audio stream, or the audio stream was not received by Asterisk. The problem may have many different reasons, but often it is a network-related issue.Le 1..
I am searching a way to dial a SIP peer, and if it does not answer within 20 seconds, play an announcement to the caller. This means that the caller would hear a ring tone for 20 seconds, and only then hear the announcement if the callee did not answe..