Jonathan H
2016-May-09 17:18 UTC
[asterisk-users] Switching between Music on Hold streams. [13.8.2]
Thanks Joshua and everyone,
Joshua's solution seems a lot simpler and works well. Only one thing
now - The reason I named the classes as I did, was so that I could
select the class based on callerID plus extension.
Unless I've misread it, I'm limited to 9 switchable classes via the
"digit=#" option, is that correct?
Or is there a clever hack around this?
extensions.conf
[streamdemo]
exten => s,1,Answer
exten => s,2,BackGround(menu)
exten => s,3,WaitExten
exten => _[2,3,4,5],1,MusicOnHold(${CALLERID(name)}${EXTEN})
;exten => s,5,Goto(s,2)
exten => _[X,t,i],1,Goto(streamdemo,s,2)
and in musiconhold.conf (4 is commented out as it's AAC and I've not
figured that one out yet - bonus points to someone who can point the
way!)
[streamdemo2]
mode=custom
digit=2
application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s
http://185.14.85.162:8020
[streamdemo3]
mode=custom
digit=3
application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s
http://stream.acbradio.org:8000/mainstream.mp3
;[streamdemo4]
;mode=custom
;digit=4
;application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s
http://199.180.75.27:80/
;http://www.mushroomfm.com/media/listen.pls
[streamdemo5]
digit=5
mode=custom
application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s
http://206.225.87.121:8000/
On 9 May 2016 at 18:00, A J Stiles <asterisk_list at earthshod.co.uk>
wrote:> On Monday 09 May 2016, Jonathan H wrote:
>> ..... {stuff deleted} .....
>> [streamdemo]
>> exten => s,1,Answer
>> exten => s,2,BackGround(menu)
>> exten => s,3,WaitExten
>> exten => s,4,Goto(s,2)
>> exten =>
>> _[2,3,4,5],1,Dial(Local/${EXTEN}@play-radio,,G(play-radio^${EXTEN}^2))
>> exten => _[2,3,4,5],2,Goto(s,2)
>
> You have an error in your dialplan! The pattern _[2,3,4,5] will match any
of
> 2, a comma, 3, a comma (again), 4, a comma or 5.
>
> I think you might mean _[2345] which will match any of 2, 3, 4 or 5 (but
> not a comma), and contains no tautologies.
>
>
> --
> AJS
>
> Note: Originating address only accepts e-mail from list! If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
> --
> _____________________________________________________________________
> -- 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
Joshua Colp
2016-May-09 17:22 UTC
[asterisk-users] Switching between Music on Hold streams. [13.8.2]
Jonathan H wrote:> Thanks Joshua and everyone, > > Joshua's solution seems a lot simpler and works well. Only one thing > now - The reason I named the classes as I did, was so that I could > select the class based on callerID plus extension. > > Unless I've misread it, I'm limited to 9 switchable classes via the > "digit=#" option, is that correct?That's correct I'm afraid. If you need even more, then without modifying the code that option won't work for you. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
Jonathan H
2016-May-09 22:10 UTC
[asterisk-users] Switching between Music on Hold streams. [13.8.2]
Thanks Joshua, I realise mine is a bit of a niche case, so I certainly don't expect to be catered for "out of the box" as it were, but is there some way Dovid's solution could be wrangled into doing what I require then? Dovid mentions this (lower case!) g option, but call me thick, but I'm not sure how it's supposed to work. I'm not being lazy here, but I really just can't figure it out! I've tried a few other things, but it really doesn't seem like the second channel can "hear" or respond to the DTMF tones. It doesn't have to be an elegant solution, any fudge will do! It did just occur to me that I could use this option I just noticed: "H - Allow the calling party to hang up by sending the DTMF sequence defined for disconnect in features.conf." Will that hang up the whole call, or just the extension which was dialled by the Dial(PJSIP/...) application? Does that make sense? (It's getting late here!). Thanks! On 9 May 2016 at 18:22, Joshua Colp <jcolp at digium.com> wrote:> Jonathan H wrote: >> >> Thanks Joshua and everyone, >> >> Joshua's solution seems a lot simpler and works well. Only one thing >> now - The reason I named the classes as I did, was so that I could >> select the class based on callerID plus extension. >> >> Unless I've misread it, I'm limited to 9 switchable classes via the >> "digit=#" option, is that correct? > > > That's correct I'm afraid. If you need even more, then without modifying the > code that option won't work for you. > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.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
Dovid Bender
2016-May-11 10:09 UTC
[asterisk-users] Switching between Music on Hold streams. [13.8.2]
If you ever figure out AAC in Asterisk for MOH let me know. The ones that I have working is MP3 and MMS. On Mon, May 9, 2016 at 1:18 PM, Jonathan H <lardconcepts at gmail.com> wrote:> Thanks Joshua and everyone, > > Joshua's solution seems a lot simpler and works well. Only one thing > now - The reason I named the classes as I did, was so that I could > select the class based on callerID plus extension. > > Unless I've misread it, I'm limited to 9 switchable classes via the > "digit=#" option, is that correct? > > Or is there a clever hack around this? > > extensions.conf > > [streamdemo] > exten => s,1,Answer > exten => s,2,BackGround(menu) > exten => s,3,WaitExten > exten => _[2,3,4,5],1,MusicOnHold(${CALLERID(name)}${EXTEN}) > ;exten => s,5,Goto(s,2) > exten => _[X,t,i],1,Goto(streamdemo,s,2) > > and in musiconhold.conf (4 is commented out as it's AAC and I've not > figured that one out yet - bonus points to someone who can point the > way!) > > [streamdemo2] > mode=custom > digit=2 > application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s > http://185.14.85.162:8020 > > [streamdemo3] > mode=custom > digit=3 > application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s > http://stream.acbradio.org:8000/mainstream.mp3 > > ;[streamdemo4] > ;mode=custom > ;digit=4 > ;application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s > http://199.180.75.27:80/ > ;http://www.mushroomfm.com/media/listen.pls > > [streamdemo5] > digit=5 > mode=custom > application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s > http://206.225.87.121:8000/ > > On 9 May 2016 at 18:00, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > > On Monday 09 May 2016, Jonathan H wrote: > >> ..... {stuff deleted} ..... > >> [streamdemo] > >> exten => s,1,Answer > >> exten => s,2,BackGround(menu) > >> exten => s,3,WaitExten > >> exten => s,4,Goto(s,2) > >> exten => > >> _[2,3,4,5],1,Dial(Local/${EXTEN}@play-radio,,G(play-radio^${EXTEN}^2)) > >> exten => _[2,3,4,5],2,Goto(s,2) > > > > You have an error in your dialplan! The pattern _[2,3,4,5] will match > any of > > 2, a comma, 3, a comma (again), 4, a comma or 5. > > > > I think you might mean _[2345] which will match any of 2, 3, 4 or 5 > (but > > not a comma), and contains no tautologies. > > > > > > -- > > AJS > > > > Note: Originating address only accepts e-mail from list! If replying > off- > > list, change address to asterisk1list at earthshod dot co dot uk . > > > > -- > > _____________________________________________________________________ > > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160511/a1123cf9/attachment.html>
Jonathan H
2016-Aug-12 14:23 UTC
[asterisk-users] Switching between Music on Hold streams. [13.8.2]
Hello! I thought having finally "cracked it", I might as well post what I've done. https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/moh-switching.md Can someone please take a quick look and see if there's anything I could have done better or more efficiently, or if anything stands out as particularly horrific? Basically, it uses an app called crudini to add sections to musiconhold.conf, then does an moh reload. When the user has finished listening and presses * then the remote extension is dropped and the caller returns to the current menu. The nice thing about this is that even if two callers call and listen to the same moh stream, when one hangs up, even though it deletes the config and reloads moh, Asterisk is nice to the other caller and they keep listening. The end result is what I wanted, which is to not have any extra CPU load or network usage when no-one is listening. And if more than one person is listening, it's still only "using" one remote stream, as I've uncommented the cachertclasses value. ---------------- [general] cachertclasses=yes ; use 1 instance of moh class for all users who are using it ---------------- As a little bonus, I've put what I think is a clever little "menu maker" in, which grabs and caches short audio files using the free plan from voicerss.org. If anyone wants to try it in practice, call UK +44 20 36 37 60 70 - this number is working as of the 12th of August and I'll leave it up at least over the weekend, but if you're reading this in a few weeks, don't expect it to work! (I'm allowed to use these streams before anyone panics!). On 11 May 2016 at 11:09, Dovid Bender <dovid at telecurve.com> wrote:> If you ever figure out AAC in Asterisk for MOH let me know. The ones that > I have working is MP3 and MMS. > > On Mon, May 9, 2016 at 1:18 PM, Jonathan H <lardconcepts at gmail.com> wrote: > >> Thanks Joshua and everyone, >> >> Joshua's solution seems a lot simpler and works well. Only one thing >> now - The reason I named the classes as I did, was so that I could >> select the class based on callerID plus extension. >> >> Unless I've misread it, I'm limited to 9 switchable classes via the >> "digit=#" option, is that correct? >> >> Or is there a clever hack around this? >> >> extensions.conf >> >> [streamdemo] >> exten => s,1,Answer >> exten => s,2,BackGround(menu) >> exten => s,3,WaitExten >> exten => _[2,3,4,5],1,MusicOnHold(${CALLERID(name)}${EXTEN}) >> ;exten => s,5,Goto(s,2) >> exten => _[X,t,i],1,Goto(streamdemo,s,2) >> >> and in musiconhold.conf (4 is commented out as it's AAC and I've not >> figured that one out yet - bonus points to someone who can point the >> way!) >> >> [streamdemo2] >> mode=custom >> digit=2 >> application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s >> http://185.14.85.162:8020 >> >> [streamdemo3] >> mode=custom >> digit=3 >> application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s >> http://stream.acbradio.org:8000/mainstream.mp3 >> >> ;[streamdemo4] >> ;mode=custom >> ;digit=4 >> ;application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s >> http://199.180.75.27:80/ >> ;http://www.mushroomfm.com/media/listen.pls >> >> [streamdemo5] >> digit=5 >> mode=custom >> application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s >> http://206.225.87.121:8000/ >> >> On 9 May 2016 at 18:00, A J Stiles <asterisk_list at earthshod.co.uk> wrote: >> > On Monday 09 May 2016, Jonathan H wrote: >> >> ..... {stuff deleted} ..... >> >> [streamdemo] >> >> exten => s,1,Answer >> >> exten => s,2,BackGround(menu) >> >> exten => s,3,WaitExten >> >> exten => s,4,Goto(s,2) >> >> exten => >> >> _[2,3,4,5],1,Dial(Local/${EXTEN}@play-radio,,G(play-radio^${EXTEN}^2)) >> >> exten => _[2,3,4,5],2,Goto(s,2) >> > >> > You have an error in your dialplan! The pattern _[2,3,4,5] will match >> any of >> > 2, a comma, 3, a comma (again), 4, a comma or 5. >> > >> > I think you might mean _[2345] which will match any of 2, 3, 4 or 5 >> (but >> > not a comma), and contains no tautologies. >> > >> > >> > -- >> > AJS >> > >> > Note: Originating address only accepts e-mail from list! If replying >> off- >> > list, change address to asterisk1list at earthshod dot co dot uk . >> > >> > -- >> > _____________________________________________________________________ >> > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160812/54fdaa27/attachment.html>