similar to: MoH issue

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

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

2010/10/24 Olivier <oza_4h07 at yahoo.fr> > > > 2010/10/14 Danny Nicholas <danny at debsinc.com> > >> ------------------------------ >> >> *From:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Olivier >> >> *Sent:* Thursday, October 14, 2010 3:34 PM >> *To:*

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

Dears -How can I stop MOH when status of the dial is ringing and let the user hear the Ring Back Tone from the termination Gateway. Even I can see in the CLI debugging the status is ringing -my idea is to add music on hold stop when asterisk detect -- SIP/OPNS-096456c0 is ringing line In which script this line located? -- Executing [97130245612 at default:1]

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 =>

Trying to test the music on hold function and can't seem to get it to work. If anyone has it running, could you give me a clue? (I have googled and found lots of questions, but no real suggestions.) I downloaded and installed the mpg123 package. From the RH9 console I can start the executable and hear the music via the speakers. The executable is located in /usr/bin. (That works!) I set the

Yea, I have faked that with a silent mp3, but to do it right it should also be a config flag in the agent.conf file for each agent, prolly add another arg to each agent definition for the MOH class, & the arg 'none' means don't play music for that agent -----Original Message----- From: James Golovich <james@wwnet.net> To: asterisk-users@lists.digium.com

This message was just bounced back to me. I am not sure if it made it to the list originally or not, as I received no responses. Since this message was written, I have installed Zap hardware into this server. The Zap channels can be transferred to the Meetme conference. The IAX2 calls still cannot. Any suggestions will be greatly appreciated. Sincerely, Trevor Hammonds Trevor G.

I have a peculiar situation, and am hoping someone on the list can offer assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

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

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 >

Hi Guys, I've attempted to get this moh-native thing to work with no success. I've reviewed wiki, mantis and e-mail postings and I'm confused. The latest I've read is native moh should be in asterisk-addons in format_mp3, but what version will it work with? I've tried asterisk 1.0.1, 1.0.5, addons 1.0.1, 1.0.4 and also -r stable CVS. I followed the wiki example with

Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to a queue without option '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).

Is is possible to specify the MOH Class when defining a MeetMe extension? I tried exten => 300,1,MeetMe(300|M(class)) But that did not work. Thx, -Rob.

I just installed the newly released Asterisk 1.4.7 and I cannot get music on hold. I am using the default settings with the wav files. Here is what I get on the cli from any sip phone: -- Executing [7001 at oficina:1] NoCDR("SIP/1120-084e6010", "") in new stack -- Executing [7001 at oficina:2] Answer("SIP/1120-084e6010", "") in new stack --

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

All, I am using Asterisk 1.4.4 and it is not playing any MOH. I think the underlying problem is the following error: [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/moh/asterisk' [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player Now it does not matter what I change in the

If i do an asterisk -rx "moh reload" MoH stops and restarts on exsisting channels. If I do an moh reload through the Manager Interface Sound is dead on exsisting channels. Any other idea for an workaround? Hi, I have problems with 1.2.14 and musiconhold.conf and realtime. I have to do moh reload at CLI to use the classes stored in mysql. Otherwise nothing is found if using