Re: [asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?
On Friday 01 November 2019 at 22:29:28, Dan Cropp wrote: > We have a customer who wants us to record anywhere from 2-4 participants on > a call in stereo (as opposed to mono) quality audio. I'm assuming you mean you want to get one stereo recording for each participant, where the left channel is the participant and the right channel is the rest of the conference? If that's not correct, what do you want the two channels of a stero recording to contain? > We are using asterisk 16.6.1 > We are also currently using AMI/AsyncAGI and ConfBridge to bring the > parties together. I believe recording in the various file formats (based > on extension), it's always recording in mono quality. > > My one thought is to transition to using ARI Bridge (instead of ConfBridge) > and streamed audio using ExternalMedia. Then have a media server capture > the external media packets, stripping the payload information and write > directly to a file. Would that audio be of ulaw stereo or mono? Suppose it *is* stereo - what would you expect the two channels to contain? It sounds like you want one single stereo recording of a conference with multiple participants. Are they all using stereo telephones and generating two-channel audio into the conference - or what?? > Any suggestions? How about a simple MixMonitor with btr options on each participant who dials in, before they get placed into the conference? Then the t channel should be the participant and the r channel should be the rest of the conference. Regards, Antony. -- What do you call a dinosaur with only one eye? A Doyouthinkesaurus. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?
We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one thought is to transition to using ARI Bridge (instead of ConfBridge) and streamed audio using ExternalMedia. Then have a media server capture the external media packets, stripping the payload information and write directly to a file. Would that audio be of ulaw stereo or mono? Any suggestions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stuck "channel"
On Thu, Oct 31, 2019 at 11:05 PM Carlos Chavez wrote: > I have tried both by hand and hitting tab to auto complete: > > *CLI> channel request hangup Message/ast_msg_queue > Message/ast_msg_queue is not a known channel > This channel is used for processing all out of dialog SIP MESSAGE requests in the dialplan. It cannot be hung up. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Own MOH incorrectly kicking in instead of the MOH of the callee
Hello all! I'm reproducibly getting my *own MOH* if I should get the MOH of the Callee instead. I can see this with asterisk 13 and 16 (and probably before, too). The reason of the wrong MOH is an in dialog reInvite received from trunk, which sends a SDP containing a=sendonly After this reInvite, I can hear own MOH instead of the MOH of the Caller. The situation is cleared by another reInvite received from the trunk containing a=sendrecv Is this expected behavior? I don't think it should act like this. BTW: I'm additionally using FreePBX. Maybe it's a problem of FreePBX? Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
