search for: musicclass

Hi Guys I wonder if its possible to set a different MoH based on groups, I mean if one of the Admin group put on hold the call play music 1, if another from Technical Support put on hold the call play music 3, something like this Admin - Music1 Contrallors - Music 2 Technical Support - Music 3 Thanks -- -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141005/03df3f3b/attachment.html>

...But not able to use MOH. I make simple extension in asterisk conf file but no success :( Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf* [incoming] exten => 6000,1,Answer exten => 6000,n,Set(CHANNEL(musicclass)=BSNL) exten => 6000,n,Set(foo=${CHANNEL(musicclass)}) exten => 6000,n,MusicOnHold(BSNL) * 2) Musiconhold.conf* [BSNL] mode=files directory=/var/lib/asterisk/moh/MTNL [Bhati] mode=files directory=/var/lib/asterisk/moh/Bhati *3) Error log:-* == Using SIP RTP TOS bits 184 == Using SIP RTP...

...file but no success :( But when I used Vanilla Asterisk then All things are working.... Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf* [incoming] exten => 6000,1,Answer exten => 6000,n,Set(CHANNEL(musicclass)=BSNL) exten => 6000,n,Set(foo=${CHANNEL(musicclass)}) exten => 6000,n,MusicOnHold(BSNL) * 2) Musiconhold.conf* [BSNL] mode=files directory=/var/lib/asterisk/moh/MTNL [Bhati] mode=files directory=/var/lib/asterisk/moh/Bhati *3) Error log:-* == Using SIP RTP TOS bits 184 == Using SIP RTP...

I am having a strange problem with MOH. Say I have two users, A and B. I can set MOH in the extension for B and if A calls B and B hits hold, A will hear B's hold music. If however A hits hold, it goes to the default music. If I pull the setmusiconhold from extensions.conf and use musicclass in sip.conf under the peer A, I get the same thing. Peer A has musicclass set and A calls B and B hits hold, A hears their own music that was specified and B hears the default. I can't find any way to call someone, put them on hold and have them hear my music, I can hear what they specify, bu...

Hello list, what musiconhold class has priority : - field "musiconhold" of the SIPaccount and field "musiconhold" of a queue or - Set(CHANNEL(musicclass)=) ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110201/b27a0534/attachment.htm>

...re just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within the same login, socket session. Just trying to add this additional task. This is from PHP as you may have recognized. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Thanks in advance for any help on this.# Set the Music on Hold fputs($socket2, "Action: Setvar\r\n"); fputs($socket2, "Channel: ".$channel."\r\n&quot...

...,Goto(incoming1,s,1) [from-pstn8] exten => s,1,Set(FROM_DID="6780000000) exten => s,n,NoOp(id is ${FROM_DID}) exten => s,n,Goto(incoming1,s,1) [incoming1] include => from-internal include => parkedcalls exten => s,1,Answer exten => s,n,Wait(1) exten => s,n,Set(CHANNEL(musicclass)=QCI) exten => s,n,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=10) exten => s,n,Background(thank-you-for-calling) exten => s,n,Dial(${QPHONE0}&${QPHONE1}&${QPHONE2}&${QPHONE3}&${QPHONE4}&${QPHONE5}&$ {QPHONE6}&${QPHONE7},40,Ttr) exten => s,n,...

...# Call comes in here and is answered -- SIP/snom360-00000d6f answered DAHDI/2-1 -- Executing [s at macro-set-moh-call:1] GotoIf("SIP/snom360-00000d6f", "0?done") in new stack -- Executing [s at macro-set-moh-call:2] Set("SIP/snom360-00000d6f", "CHANNEL(musicclass)=m-default") in new stack -- Executing [s at macro-set-moh-call:3] NoOp("SIP/snom360-00000d6f", "") in new stack ### Here the call is being blind transferred to the Park number -- Started music on hold, class 'default', on DAHDI/2-1 -- Stopped music on...

...: [Feb 26 12:35:19] NOTICE[1143][C-0000025d]: app_dial.c:958 do_forward: Not accepting call completion offers from call-forward recipient Local/1XXXXXXXXXX at default-00000013;1 So I put this in my dial plan: 1AAAAAAAAAA => { NoOp(${CALLERID(num)}); Ringing; Set(CHANNEL(musicclass)=none); Dial(${OUTBOUND-TRUNKR}/1XXXXXXXXXX,30); Voicemail(198,u); }; So basically as soon as someone calls incoming number AAAAAAAAAA, Asterisk dials phone number XXXXXXXXXX. it's a quick and dirty way to call forward.. and this does the same thing, there's a good 8 secon...

...s are directed to?all of my?exensions? extensions.conf [from-myprovider] exten => *DID number*,1,Answer exten => *DID number*,2,Dial(SIP/1000) exten => *DID number*,3,Queue(support) ;not sure if this line belongs here exten => *DID number*,4,Hangup ? queues.conf ? [general] [support] musicclass=default strategy=rrmemory joinempty=no leavewhenempty=yes ringinuse=no Member => SIP/1000 Member => SIP/1001 agent => 1000,1000 agent => 1001,1001 ? When using the current config the caller will listen to the 'music on hold' until the agent answers but calls are only being forw...

Hello list ! I have some configuration issues with queues, but I'm sure they are minor and for someone who has already configured queues it could be trivial. This is my queue configuration : [VC_support_queue] musicclass = default strategy = ringall timeout = 20 retry = 5 wrapuptime=15 autofill=yes autopause=no maxlen = 0 setinterfacevar=yes announce-frequency = 0 periodic-announce-frequency=0 announce-holdtime = no ; announce-round-seconds = 10 ; queue-thankyou= ;queue-youarenext = queue-youarenext ;queue-thereare...

...tions) exten => s,2,GotoIf($["${SCREEN_OPT}" = "" ]?s,1) exten => s,3,GotoIf($["${SCREEN_OPT}" = "3" ]?3,1) exten => 3,1,Set(MACRO_RESULT=CONTINUE) exten => t,1,Set(MACRO_RESULT=BUSY) exten => h,1,Set(MACRO_RESULT=BUSY) queues.conf [mainqueue] musicclass = default strategy = ringall timeout = 600 joinempty = yes member => local/306 at screen ~Shaun

I have a requirement to play different hold messages depending upon the extension that originated the call. I noticed a musicclass setting in sip.conf, but it appears this is global. I tried setting this on all of my individual extensions, but it didn't have any affect. Is there a way to achieve this, either through sip.conf or in the dial plan? Thanks, James

...= asterisk keepalive = 60 debug = 10 context = from-sccp-internal dateFormat = M/D/YA port = 2000 disallow=all allow=alaw allow=ulaw allow=g723 firstdigittimeout = 60 digittimeout = 8 autoanswer_ring_time = 0 autoanswer_tone = 0x32 remotehangup_tone = 0x32 transfer_tone = 0 callwaiting_tone = 0x2d musicclass=default language=en rtptos = 184 echocancel = on silencesuppression = off trustphoneip = no tos = 0x68 [devices] type = 7960 autologin = 2002 description = phone2002 dtmfmode = inband imageversiom = P00307020200 dnd = on trustphoneip = no speeddial = 2000 private = on device => SEP00036BC3...

...-frequency=15 periodic-announce=/var/lib/asterisk/sounds/ogm/Hold-Ran-1,/var/lib/asterisk/sounds/ogm/Hold-Ran-2,/var/lib/asterisk/sounds/ogm/Hold-Ran-3,/var/lib/asterisk/sounds/ogm/Hold-Ran-4 strategy=rrmemory joinempty=yes retry=5 announce-holdtime=no wrapuptime=15 announce-frequency=60 timeout=10 musicclass=default autofill=no ringinuse=no Any ideas? Thanks, Mike Clark

In my context I have: exten => _NXXXXXX,1,Set(CHANNEL(musicclass)=default) exten => _NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav) exten => _NXXXXXX,n,MixMonitor(${recordfilename},b) but the recorded conversation ended in 10min so it = 600sec I was looking in asterisk configuration file for "600...

...2592603325 at default:1] Verbose("SIP/192.168.20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") in new stack == PROXY Call from 0123456 to 000972592603325 -- Executing [000972592603325 at default:2] Set("SIP/192.168.20.120-0000002a", "CHANNEL(musicclass)=default") in new stack -- Executing [000972592603325 at default:3] GotoIf("SIP/192.168.20.120-0000002a", "0?dialluca") in new stack -- Executing [000972592603325 at default:4] GotoIf("SIP/192.168.20.120-0000002a", "0?dialfax") in new stack -...

...com 501 phones and I'm adding agents to the queues using Addqueuemember(). I believe I have the call limits and limitonpeer settings right in sip.conf. The only difference between the two queues is one has a higher weight. Any suggestions would be greatly appreciated. [our-support-queue] musicclass = default strategy = leastrecent timeout = 12 retry = 15 wrapuptime=30 weight=0 autopause=yes maxlen=0 joinempty=strict leavewhenempty=strict ringinuse=no context=queue-out periodic-announce-frequency=60 announce-holdtime=no periodic-announce=my-prompt-29 Thanks, Nick

...ption 'r' specified - moh plays as expected. Now, when that call is answered, all is fine. Trouble comes when that person then puts the caller on-hold. No moh is heard by the caller (in fact, they get silence). If I use 'r' - then ringing is heard - but the queue's musiconhold/musicclass is ignored completely. When the caller is put on hold, they do hear moh but the default moh context is used - not the moh of the queue. What I need is for the queue's moh to be used when the caller is put on hold (and without using the 'r' feature). Is this possible? * 1.8.16.0 (tri...