Re: [asterisk-users] Setting channel musicclass from AGI
On Sun, Oct 5, 2014 at 6:40 PM, James Lamanna wrote: > Hi, > Since SetMusicOnHold() is being deprecated, how do we set the channel > musicclass from an AGI script? > Last time I checked you can't call dialplan functions from AGI. > Actually, you can. Any time you can evaluate or set a channel variable, you can also evaluate or set a dialplan function. Hence, you can use both 'get variable' [1] or 'set variable' [2]. You could also use 'exec' and call the Set dialplan application directly. [1] https://wiki.asterisk.org/wiki/display/AST/AGICommand_get+variable [2] https://wiki.asterisk.org/wiki/display/AST/AGICommand_set+variable -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting channel musicclass from AGI
Hi, Since SetMusicOnHold() is being deprecated, how do we set the channel musicclass from an AGI script? Last time I checked you can't call dialplan functions from AGI. Thanks. -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how can queue agents choose which call to answer?
... and to continue my thought, if nothing else is possible, would it be a Very Bad Idea to just delete the ABANDON log (queue_log goes to mysql via odbc) automatically after it's created? In h extension? -- marie On 05.10.2014, at 20:42, Marie Fischer wrote: > Thanks for your ideas. I set up a solution via AMI Redirect and it works > nicely. > > The only question now is queue metrics, as you also mentioned - the > redirected calls get logged as ABANDON in the queue log. I could of course > add a custom entry to the log via QueueLog function to show the call was > actually redirected, but is there a way to disable/change the ABANDON log > itself? It seems from this discussion FOP has the same problem: > http://forum.fop2.com/1746-call-pickup-function-causes-dummy-entry-in-cdr-database/0 > > -- > > marie > > On 23.09.2014, at 22:02, Scott Griepentrog wrote: > >> You can use any number of methods for redirecting a call from the queue to a >> specific agent. These include off the shelf products such as FOP or >> iSymphony, or even something custom built that can display calls and direct >> Asterisk (usually through AMI) to transfer the call to a new destination. >> >> However, you will need to be aware that your queue metrics may not count it >> as a normally handled call, since the call is yanked out of the queue to >> transfer directly to an agent via a separate tool. >> >> You may also want to look into building a custom queue-like solution through >> ARI, using a Stasis application to manage callers on hold in waiting >> bridges, and then delivering them to agents completely under control of your >> application. In this case you would need to create your own queue logging >> data to your metrics solution, which would allow you to record calls >> correctly even when transferred early. >> >> >> On Tue, Sep 23, 2014 at 1:41 PM, Michael Keuter >> wrote: >> >> Am 23.09.2014 um 19:49 schrieb Marie Fischer : >> >>> Hi everybody, >>> >>> I'm looking for a solution for the following scenario: >>> >>> • Asterisk queue >>> • At peak hours, there will be more callers then queue members/agents, so >>> some callers will spend some time on hold >>> • Agents should be able to choose which of the on hold calls to answer >>> instead of answering the next one in queue >>> >>> We already have a web interface where agents can see the callers on hold, >>> so the best solution would be if they could just click a callers number to >>> get his call. But I have not found a way to tell Asterisk to do something >>> to a call on hold in a queue. >>> >>> Priority queues are not really an option, as the agents will be deciding on >>> the fly which caller is more important. >>> >>> I am not really sure if queues are the correct solution for this problem. >>> However, we have existing statistics built for queue logs, so it would be >>> really nice if the solution was queue-based. >>> >>> Thanks for any thoughts, >>> >>> -- >>> >>> marie >> >> >> Hello Marie, >> >> maybe FOP2 [1] is an option for you. There you can visually "pick up" a >> call from a queue. >> It's not open source though. >> >> [1] http://www.fop2.com >> >> Michael >> >> http://www.mksolutions.info >> >> >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> -- >> >> Scott Griepentrog >> Digium, Inc · Software Developer >> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US >> direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 >> Check us out at: http://digium.com · http://asterisk.org >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.
Re: [asterisk-users] how can queue agents choose which call to answer?
Thanks for your ideas. I set up a solution via AMI Redirect and it works nicely. The only question now is queue metrics, as you also mentioned - the redirected calls get logged as ABANDON in the queue log. I could of course add a custom entry to the log via QueueLog function to show the call was actually redirected, but is there a way to disable/change the ABANDON log itself? It seems from this discussion FOP has the same problem: http://forum.fop2.com/1746-call-pickup-function-causes-dummy-entry-in-cdr-database/0 -- marie On 23.09.2014, at 22:02, Scott Griepentrog wrote: > You can use any number of methods for redirecting a call from the queue to a > specific agent. These include off the shelf products such as FOP or > iSymphony, or even something custom built that can display calls and direct > Asterisk (usually through AMI) to transfer the call to a new destination. > > However, you will need to be aware that your queue metrics may not count it > as a normally handled call, since the call is yanked out of the queue to > transfer directly to an agent via a separate tool. > > You may also want to look into building a custom queue-like solution through > ARI, using a Stasis application to manage callers on hold in waiting bridges, > and then delivering them to agents completely under control of your > application. In this case you would need to create your own queue logging > data to your metrics solution, which would allow you to record calls > correctly even when transferred early. > > > On Tue, Sep 23, 2014 at 1:41 PM, Michael Keuter > wrote: > > Am 23.09.2014 um 19:49 schrieb Marie Fischer : > > > Hi everybody, > > > > I'm looking for a solution for the following scenario: > > > > • Asterisk queue > > • At peak hours, there will be more callers then queue members/agents, so > > some callers will spend some time on hold > > • Agents should be able to choose which of the on hold calls to answer > > instead of answering the next one in queue > > > > We already have a web interface where agents can see the callers on hold, > > so the best solution would be if they could just click a callers number to > > get his call. But I have not found a way to tell Asterisk to do something > > to a call on hold in a queue. > > > > Priority queues are not really an option, as the agents will be deciding on > > the fly which caller is more important. > > > > I am not really sure if queues are the correct solution for this problem. > > However, we have existing statistics built for queue logs, so it would be > > really nice if the solution was queue-based. > > > > Thanks for any thoughts, > > > > -- > > > > marie > > > Hello Marie, > > maybe FOP2 [1] is an option for you. There you can visually "pick up" a call > from a queue. > It's not open source though. > > [1] http://www.fop2.com > > Michael > > http://www.mksolutions.info > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Scott Griepentrog > Digium, Inc · Software Developer > 445 Jan Davis Drive NW · Huntsville, AL 35806 · US > direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 > Check us out at: http://digium.com · http://asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pjsip and regcontext (for DUNDi)
Hi, Was this question not appropriate for asterisk-users maybe? Should I post in dev instead? Dan On 4 Oct 2014 15:48, "Dan Ballance" wrote: > Hi guys, > > I'm building a PoC Asterisk 12 cluster based on a number of guides I've > found on the net. The basic concept is using ARA in conjunction with DUNDi. > I have set up ARA with pjsip according to this excellent guide here: > > https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime > > This is working nicely, so now I am turning my attention to DUNDi, as per > this guide here: > > > http://www.ntegratedsolutions.com/wp-content/uploads/2012/07/Using_DUNDi_with_a_Cluster_of_Asterisk_Servers.pdf > > Its seems a really neat solution and I'm keen to implement something > similar, however I believe it was written before the pjsip channel driver > and I've hit a potential issue I think. The guides for configuring DUNDi > seem to suggest using regcontext in sip.conf: > > [general] > > regcontext=sipregistration > > However I can't seem to find an equivalent declaration for pjsip.conf. So > my questions are: > > 1) Is there a way to achieve the same functionality with pjsip? > > 2) Is DUNDi still being maintained and used? If so, then how should it be > configured with modern versions of Asterisk? > > 3) If DUNDi is not really used in modern set-ups, then what are my > alternatives? > > I really have searched and read and Googled everything I can but I can't > seem to find anything on configuring DUNDi with pjsip. Hoping one of you > people can point me in the right direction! > > many thanks in advance, > > Dan > > > > > > > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail message number off by one when using ODBC storage
... 'cause message file names start with 0 (msg.wav). -- marie On 05.10.2014, at 18:45, Leandro Dardini wrote: > Hello, > have you noticed the message num (VM_MSGNUM) is off by one? > > For example, I receive the following message: > > "Just wanted to let you know you were just left a 0:03 long message (number > 7)" > > but in attach there is the msg0006.wav > > Leandro > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail message number off by one when using ODBC storage
Hello, have you noticed the message num (VM_MSGNUM) is off by one? For example, I receive the following message: "Just wanted to let you know you were just left a 0:03 long message (number 7)" but in attach there is the msg0006.wav Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
