similar to: using Playback() to play a random sound file

I configured my sla.conf to use with a Polycom phone. I have no idea if I did it right, however, none of the console "sla" commands exist. Do I have to something special to compile in this support, or should it just work out of the box? ~jay

Hi, Is there any way to run a command (AGI script, whatever else) at that moment when the call that was in the queue is being bridged to a specific agent? An examples of what I would want to do with this is, for example, have Asterisk ask the caller for his 4 digit customer ID before being put in the Queue. Once I know who the caller is being connected to (which agent) I'd run a

Hello. Is it possible to change the defined moh sound file within an extension? I have: exten => 18,1,Answer exten => 18,n,Wait(3) exten => 18,n,SetMusicOnHold(durchwahl) exten => 18,n,Dial(SIP/118,15,m) exten => 18,n,Hangup Now i have the situation someone calls and my phone is ringing while moh (durchwahl) is playing. When i pickup the call and press the hold button during

I recently upgraded Asterisk from 1.8.x to 13.x and am now finding that music on hold isn't working like it used to. It seems that even though the correct MoH class is being set, the system still plays the "default" music. All of my call handling is done with an AGI script. When a call is made, the AGI script sets the MoH class before dialing. The log indicates that the correct

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:

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

I'd like multiple people to be able to dial in and listen to various live radio streams. I was told that the correct resource-friendly way would be to setup a MoH class, and then select that from the dialplan. This works well, but how do I switch between streams? Someone correct me if I'm wrong, but from previous similar questions a few years ago it seems like once you've entered a

Hello list, I have defined a new MoH-class in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; *[106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes* In sip.conf I have this commented out : ;mohinterpret=default ;mohsuggest=default Asterisk sees these moh-classes and files : vps2301*CLI> moh show classes Class: default Mode: files

Hi there; I didn't see any "G" option in the example above, and the usage for the option parameters is entirely undocumented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial The G options are as below G - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten

Hi, Anyone knows if there is a way to play a list of sound file in a round robin mode (at specific interval) while someone in waiting in moh in a queue? Ok, you enter a queue and wait listening to moh, every X minutes a sound file is played from a list of sound files to be played. If that possible and if so how? Thanks for any pointers. Andre

Hello. Is it possible to "mix" musiconhold music and playback voices? What i want to do is something like this: A person calls a number, gets a playback voice while in background music is playing. The configuration i use at the moment don't do what i want. Someone knows how to do it? Thanks in advance. exten => 18,1,Answer exten => 18,n,Background() exten =>

Hi all, I'm trying to make some simple MOH (Music On Hold) working. So far I've failed miserably - so I turn here for help. Basically I've been using the wiki and all the sample confs I could from there and via google. The queue system seems to work fine with my limited setup. Just 2 IAX2 clients where I keep Client B busy (by making it listen to mp3 via ext. 777) but logged into

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

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

> -----Original Message----- > From: listas iPfone [mailto:listas@ipfone.com.br] > > Somebody knows why asterisk gives me that error wile playing .mp3 files? > > The files play well but the message aperas any way: > WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of > 4 > bytes) (No such file or directory)! Listas, You might try down-sampling

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

Hi all, I have created a menu from which the caller can select several options such as being transfered to our phones and my mobile phone, meetme, etc. If the caller press an invalid option i have set it to play a message like invalid choice please try again. If the caller make three invalid choices i want the call to be disconnected. what is the best way of doing that? And finally i have set up

Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround("SIP/joa-9def", "tt-weasels") in

Thanks to this thread, we got it working too... but have a question... Once this is setup... does it stream forever, or does the stream only start when someone goes on hold/into a queue/etc? If it streams forever, at 24k... it looks like over 7GB/month in bandwidth... so we're not going to want to do that if a) it streams constantly and b) my math is correct. Thanks, Doug >