Re: [asterisk-dev] Monitor vs. MixMonitor questions
On Thu, 7 Nov 2019 at 15:24, Joshua C. Colp wrote: > On Thu, Nov 7, 2019 at 11:14 AM Steve Davies wrote: > >> Hi, >> >> So I've chosen to post this in -dev in case someone manages to come up >> with a pointer to an existing patch, because if not I imagine I will do the >> work and submit it myself. Advice, corrections and opinions are most >> welcome. >> >> I noticed that 'res_monitor' is flagged as deprecated on the basis that >> 'app_mixmonitor' supports all the same features, and this led me down a >> rat-hole which I hope to dig my way out of... >> >> - I need to record in- and out- channels separately so I can mix to a >> stereo file (YES) >> - I need to pause and un-pause (NO) >> - Solution: do a stop and start/append instead (YES) >> - I only want to mix at the very end of the bridge/call (NO) >> - The automon feature should also be deprecated or merged with automixmon >> (?) >> >> I believe that the simplest solution to all of the above it to add a >> pause/unpause feature to app_mixmonitor, and extend 'struct >> ast_mixmonitor_methods' to allow broad support for an optional pause and >> unpause operation. This would then be implemented by simply pausing whether >> or not we write to the relevant filehandles. The audiohook itself would >> continue until stopped. >> >> Perhaps there is good reason why this has not been done before? If so, >> please let me know before I go too far :) >> > > I'm not aware of any real reasons why it hasn't been done, except noone > taking the time to do so or running into that functionality or getting > around it in a different way. > > I think deprecating automon is fine, although we don't really have a good > ability to deprecate such things except to state it. They don't have > support levels and other things like modules. > > Thanks for the rapid feedback - I think I'll leave the automon vs automixmon deprecation alone for now then :) I'm guessing that res_monitor is not going to be removed any time soon based on your comments. I will on the other hand have a crack at Pause/Unpause since when googling it the question has been asked quite a few times. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Gerrit usage
On Thu, Nov 7, 2019 at 8:59 AM Tomec Martin wrote: > Hi, > > after some years I have tried to submit code to gerrit and ended with > error: > > > > [asterisk]# git review 13 > > remote: Resolving deltas: 100% (3/3) > > remote: Counting objects: 66212, done > > remote: error: branch refs/publish/13/ASTERISK-28613: > > remote: You need 'Create' rights to create new references. > > remote: User: matesstar > > remote: Contact an administrator to fix the permissions > > > > I followed the manual on wiki > https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage . Am I missing > something? > > > > Thanks, > > Martin > Check your version of "git review". IIRC it needs to be 1.27 or greater since we upgraded Gerrit. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- *George Joseph* Digium - A Sangoma Company | Software Developer | Software Engineering 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct/fax: +1 256 428 6012 Check us out at: https://digium.com ยท https://sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Gerrit usage
Hi, after some years I have tried to submit code to gerrit and ended with error: [asterisk]# git review 13 remote: Resolving deltas: 100% (3/3) remote: Counting objects: 66212, done remote: error: branch refs/publish/13/ASTERISK-28613: remote: You need 'Create' rights to create new references. remote: User: matesstar remote: Contact an administrator to fix the permissions I followed the manual on wiki https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage . Am I missing something? Thanks, Martin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Monitor vs. MixMonitor questions
On Thu, Nov 7, 2019 at 11:14 AM Steve Davies wrote: > Hi, > > So I've chosen to post this in -dev in case someone manages to come up > with a pointer to an existing patch, because if not I imagine I will do the > work and submit it myself. Advice, corrections and opinions are most > welcome. > > I noticed that 'res_monitor' is flagged as deprecated on the basis that > 'app_mixmonitor' supports all the same features, and this led me down a > rat-hole which I hope to dig my way out of... > > - I need to record in- and out- channels separately so I can mix to a > stereo file (YES) > - I need to pause and un-pause (NO) > - Solution: do a stop and start/append instead (YES) > - I only want to mix at the very end of the bridge/call (NO) > - The automon feature should also be deprecated or merged with automixmon > (?) > > I believe that the simplest solution to all of the above it to add a > pause/unpause feature to app_mixmonitor, and extend 'struct > ast_mixmonitor_methods' to allow broad support for an optional pause and > unpause operation. This would then be implemented by simply pausing whether > or not we write to the relevant filehandles. The audiohook itself would > continue until stopped. > > Perhaps there is good reason why this has not been done before? If so, > please let me know before I go too far :) > I'm not aware of any real reasons why it hasn't been done, except noone taking the time to do so or running into that functionality or getting around it in a different way. I think deprecating automon is fine, although we don't really have a good ability to deprecate such things except to state it. They don't have support levels and other things like modules. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.sangoma.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Monitor vs. MixMonitor questions
Hi, So I've chosen to post this in -dev in case someone manages to come up with a pointer to an existing patch, because if not I imagine I will do the work and submit it myself. Advice, corrections and opinions are most welcome. I noticed that 'res_monitor' is flagged as deprecated on the basis that 'app_mixmonitor' supports all the same features, and this led me down a rat-hole which I hope to dig my way out of... - I need to record in- and out- channels separately so I can mix to a stereo file (YES) - I need to pause and un-pause (NO) - Solution: do a stop and start/append instead (YES) - I only want to mix at the very end of the bridge/call (NO) - The automon feature should also be deprecated or merged with automixmon (?) I believe that the simplest solution to all of the above it to add a pause/unpause feature to app_mixmonitor, and extend 'struct ast_mixmonitor_methods' to allow broad support for an optional pause and unpause operation. This would then be implemented by simply pausing whether or not we write to the relevant filehandles. The audiohook itself would continue until stopped. Perhaps there is good reason why this has not been done before? If so, please let me know before I go too far :) Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev