[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
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 ldard...@gmail.com 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
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 tzewang.do...@gmail.com 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] 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 sgriepent...@digium.com 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 li...@mksolutions.info wrote: Am 23.09.2014 um 19:49 schrieb Marie Fischer ma...@vtl.ee: 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] 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 ma...@vtl.ee 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 sgriepent...@digium.com 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 li...@mksolutions.info wrote: Am 23.09.2014 um 19:49 schrieb Marie Fischer ma...@vtl.ee: 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.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] Setting channel musicclass from AGI
On Sun, Oct 5, 2014 at 6:40 PM, James Lamanna jlama...@gmail.com 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
