Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel
Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel
Daniel Gradecak wrote:> Hi all, > I would like to know if there is any possibility to send an event when a > call is monitored? > For both start and stop monitor. > > There is no event sent on asterisk 1.2 for that monitor case. I did not > find any changes regrding that on 1.4. Am I wrong? > Is it even possible to send an event when a monitor starts or stop ? Or > is this a bad idea. > > Regards, > Daniel > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >There are no events generated when the monitor stops and starts, but since you are implicitly recording in your dialplan one way or another you can just add a userevent step before recording and after.
Hi Anthony, are you sure the monitor is started and sotoped via the dialplan ? Anthony Francis wrote:> Daniel Gradecak wrote: > >> Hi all, >> I would like to know if there is any possibility to send an event when a >> call is monitored? >> For both start and stop monitor. >> >> There is no event sent on asterisk 1.2 for that monitor case. I did not >> find any changes regrding that on 1.4. Am I wrong? >> Is it even possible to send an event when a monitor starts or stop ? Or >> is this a bad idea. >> >> Regards, >> Daniel >> >> >> _______________________________________________ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > There are no events generated when the monitor stops and starts, but > since you are implicitly recording in your dialplan one way or another > you can just add a userevent step before recording and after. > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Anthony Francis wrote:> There are no events generated when the monitor stops and starts, but > since you are implicitly recording in your dialplan one way or another > you can just add a userevent step before recording and after.You can also start monitoring through the Manager API in which case you could also generate corresponding user events. It is also possible to map monitoring to dtmf digits in features.conf. In that case generating user events would be hard. So a better solution is probably to add events directly to res_monitor.c so that they fire automatically. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: stefan.reuter at reucon.com Jabber: stefan.reuter at reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 252 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070709/022b8235/attachment.pgp
Hi Stefan, actually you probably know i am using your java-asterisk :) Yes the best solution i found till now it was to add those events to res_monitor.c. I wonder why it was not yet done, may be there was a reason or nobody needed it yet. Anyhow this would be a cool feature that others should benefit from too. If i do the "patch" is that possible for those events to be added in next versions of asterisk ? Regards, Daniel Stefan Reuter wrote:> Anthony Francis wrote: > >> There are no events generated when the monitor stops and starts, but >> since you are implicitly recording in your dialplan one way or another >> you can just add a userevent step before recording and after. >> > > You can also start monitoring through the Manager API in which case you > could also generate corresponding user events. It is also possible to > map monitoring to dtmf digits in features.conf. In that case generating > user events would be hard. > So a better solution is probably to add events directly to res_monitor.c > so that they fire automatically. > > =Stefan > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Hey Daniel, I think adding the events would be a good idea. Just open an issue on http://bugs.digium.com/ and attach your patch there. Be sure to send a disclaimer to digium so your patch can be included in the distribution (see http://asterisk.org/developers/bug-guidelines for details).> actually you probably know i am using your java-asterisk :)and of course if you already have patches for Asterisk-Java that support your new events post it to http://jira.reucon.org referencing the digium issue. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: stefan.reuter at reucon.com Jabber: stefan.reuter at reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 252 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070711/3cbc62ef/attachment.pgp
Stefan Reuter wrote:> Hey Daniel, > > I think adding the events would be a good idea. > Just open an issue on http://bugs.digium.com/ and attach your patch > there. Be sure to send a disclaimer to digium so your patch can be > included in the distribution (see > http://asterisk.org/developers/bug-guidelines for details). > > >> actually you probably know i am using your java-asterisk :) >> > > and of course if you already have patches for Asterisk-Java that support > your new events post it to http://jira.reucon.org referencing the digium > issue. > > =Stefan > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersWTF
Anthony Francis wrote:> Stefan Reuter wrote: > >> Hey Daniel, >> >> I think adding the events would be a good idea. >> Just open an issue on http://bugs.digium.com/ and attach your patch >> there. Be sure to send a disclaimer to digium so your patch can be >> included in the distribution (see >> http://asterisk.org/developers/bug-guidelines for details). >> >> >> >>> actually you probably know i am using your java-asterisk :) >>> >>> >> and of course if you already have patches for Asterisk-Java that support >> your new events post it to http://jira.reucon.org referencing the digium >> issue. >> >> =Stefan >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > WTF > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >Hi, I guess I should clarify. My name is Anthony, I was the one that said I had written a patch, if Daniel also said he had done so and I missed that email I apologize. Anthony
Dear Sir, On Wed, 11 Jul 2007, Anthony Francis wrote:> WTFI am intrigued by your ideas and would like to subscribe to your quarterly newsletter. -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671
Anthony Francis wrote:> I guess I should clarify. My name is Anthony, I was the one that said I > had written a patch, if Daniel also said he had done so and I missed > that email I apologize.Well then disclaim it and post it to the asterisk bug tracker or post its issue id if you already did. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: stefan.reuter at reucon.com Jabber: stefan.reuter at reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 252 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070711/767e5567/attachment.pgp