Hi all, Continuing a previous post regarding background music during a call, there is a strange miserable problem which I am unable to understand. It works fine on some systems as I have tested it with sip and with dahdi + pri signalling with digium hardware on one of my production server, but when I use it on my other production server running digium hardware + ss7 signalling, it often looses music voice when muting /unmuting or decreasing / increasing volume. and music never comes back. some times music stops automatically after a couple of seconds and does not resume by any mean.
below is the email chain for background music. any help? ------------------------------------------------------------------------------------------------------------------------------------------------------ [asterisk-users] Background music during a call Rizwan Hisham rizwanhasham at gmail.com Tue May 10 10:59:36 CDT 2011 Previous message: [asterisk-users] Background music during a call Next message: [asterisk-users] Background music during a call Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Ooops, here is the correct version, Missed the capital X option in meetme before which lets you control the volume etc. [chat-room] exten => love,1,Goto(love-a,1) exten => love,2,Goto(love-b,1) exten => love-a,1,Set(__MOH=love) exten => love-a,n,Dial(Local/fake at chat- room,,G(chat-room,chat,1)) exten => love-b,1,Goto(chat,100) exten => curse,1,Goto(curse-a,1) exten => curse,2,Goto(curse-b,1) exten => curse-a,1,Set(__MOH=curse) exten => curse-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1)) exten => curse-b,1,Goto(chat,100) exten => fake,1,Answer exten => fake,2,MusicOnHold(${MOH}) exten => chat,1,Goto(100) exten => chat,2,MeetMe(${MM},dx1qX) exten => chat,100,MeetMe(${MM},daAx1qX) exten => h,1,MeetMeAdmin(${MM},K) exten => 4,1,MeetMeAdmin(${MM},t,2) exten => 6,1,MeetMeAdmin(${MM},T,2) exten => 2,1,MeetMeAdmin(${MM},M,2) exten => 8,1,MeetMeAdmin(${MM},m,2) exten=> _X,2,Goto(chat-room,chat,100) On Tue, May 10, 2011 at 9:57 PM, Rizwan Hisham <rizwanhasham at gmail.com>wrote: > Very nice Loan. Here is the chat-room dialplan with a little tweek which > lets you set the volume up/down or mute/unmute the song. > > Use 4/6 to increase/decrease the volume and 2/8 to mute/unmute the song > > > [chat-room] > exten => love,1,Goto(love-a,1) > exten => love,2,Goto(love-b,1) > > exten => love-a,1,Set(__MOH=love) > exten => love-a,n,Dial(Local/fake at chat- > room,,G(chat-room,chat,1)) > > exten => love-b,1,Goto(chat,100) > > exten => curse,1,Goto(curse-a,1) > exten => curse,2,Goto(curse-b,1) > > exten => curse-a,1,Set(__MOH=curse) > exten => curse-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1)) > > exten => curse-b,1,Goto(chat,100) > > exten => fake,1,Answer > exten => fake,2,MusicOnHold(${MOH}) > > exten => chat,1,Goto(100) > exten => chat,2,MeetMe(${MM},dx1q) > > exten => chat,100,MeetMe(${MM},daAx1q) > > exten => h,1,MeetMeAdmin(${MM},K) > > exten => 4,1,MeetMeAdmin(${MM},t,2) > exten => 6,1,MeetMeAdmin(${MM},T,2) > exten => 2,1,MeetMeAdmin(${MM},M,2) > exten => 8,1,MeetMeAdmin(${MM},m,2) > > exten=> _X,2,Goto(chat-room,chat,100) > > Here channel 2 always seem to be the one playing the MOH, thats why its > hard coded into the MeetMeAdmin application. > > If there is a another way to know which channel is playing the song then > please do let me know. > > Cheers > > > > On Tue, May 10, 2011 at 9:33 AM, Rizwan Hisham <rizwanhasham at > gmail.com>wrote: > >> Thanks a lot loan. Will try it today. >> >> Cheers >> >> >> On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias <indreias at gmail.com> wrote: >> >>> Updated dialplan: fix a typo when using MOH variable and now you have >>> truly dynamic conference rooms. >>> >>> Have fun, >>> Ioan. >>> >>> +++++++++++++++++++++++++++++++++++++++++ >>> exten => _[12]XXX,1,Set(__MM=${EPOCH}) >>> exten => _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1)) >>> exten => _2XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1)) >>> >>> [chat-room] >>> exten => love,1,Goto(love-a,1) >>> exten => love,2,Goto(love-b,1) >>> >>> exten => love-a,1,Set(__MOH=love) >>> exten => love-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1)) >>> >>> exten => love-b,1,Goto(chat,100) >>> >>> exten => curse,1,Goto(curse-a,1) >>> exten => curse,2,Goto(curse-b,1) >>> >>> exten => curse-a,1,Set(__MOH=curse) >>> exten => curse-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1)) >>> >>> exten => curse-b,1,Goto(chat,100) >>> >>> exten => fake,1,Answer >>> exten => fake,2,MusicOnHold(${MOH}) >>> >>> exten => chat,1,Goto(100) >>> exten => chat,2,MeetMe(${MM},dx1q) >>> >>> exten => chat,100,MeetMe(${MM},daAx1q) >>> >>> exten => h,1,MeetMeAdmin(${MM},K) >>> +++++++++++++++++++++++++++++++++++++++++ >>> >>> On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias <indreias at gmail.com> >>> wrote: >>> > I have tested the following dialplan and it could be used as a >>> > starting point. What you have to resolve is how to generate different >>> > MeetMe conference room - in the example we have only one room = 1234 >>> > >>> > If you prefix the dialled extension with 1 => you will have a "lovely >>> > chat". With 2 -> "cursing chat". >>> > >>> > HTH, >>> > >>> > Ioan >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >>> >> >> >> >> -- >> Best Ragards >> Rizwan Qureshi >> VoIP/Asterisk Engineer >> Axvoice Inc. >> >> V: +92 (0) 3333 6767 26 >> E: rizwanhasham at gmail.com >> W: www.axvoice.com >> >> > > > -- > Best Ragards > Rizwan Qureshi > VoIP/Asterisk Engineer > Axvoice Inc. > > V: +92 (0) 3333 6767 26 > E: rizwanhasham at gmail.com > W: www.axvoice.com > > -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: rizwanhasham at gmail.com W: www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110510/4a39a49c/attachment.htm> Previous message: [asterisk-users] Background music during a call Next message: [asterisk-users] Background music during a call Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] More information about the asterisk-users mailing list -- _____________________________________________________________________ -- 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