similar to: IAX vs SIP (music on hold)

Is it true music on hold isnt supported in IAX/2? I check the docs and it doesnt show a configuration setting in IAX.conf and when I put someone on hold they dont hear the music and * doesnt start the music on hold. If it doesnt is there a way to make this work? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Im new to Astererisk. I compiled the latest CVS and setup the server. It looks like things are working. I'm running kphone, x-lite and sjphone to test things out. The kphone (local to the asterisk server) can call and receive calls from any of the 2 windows machines. The first windows phone I start I can send/receve calls the second one I cannot. I. No matter which one I start first only

As posted on blogs.digium.com today: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ the Asterisk project has changed providers for Music-On-Hold (MOH) content distributed with/for Asterisk. In addition to the change for future Asterisk releases, we have also opted to rebuild historical releases with the new MOH content, in an effort to eliminate unnecessary distribution of

As posted on blogs.digium.com today: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ the Asterisk project has changed providers for Music-On-Hold (MOH) content distributed with/for Asterisk. In addition to the change for future Asterisk releases, we have also opted to rebuild historical releases with the new MOH content, in an effort to eliminate unnecessary distribution of

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hmm, My music on hold has always worked fine. But I discovered that under incoming IAX2 calls they don't get any MOH! All I could find was a comment saying let me know if you find a solution... Nor does the debugger does say: Started music on hold So it's not starting the MOH, why? I do have it configured and it does play under other

Can someone tell me how Asterisk handles music-on-hold between servers? Documentation for this is non-existent. Lets say user A, who is registered on pbx1, calls user B, who is registered on pbx2. 1. User A puts user B on hold. The moh that is played to user B should be specified according to user A. Which pbx box should this be set on? pbx1? pbx2? Both? 2. Is the situation any different if the

Hi all, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see

Hi all, I've asterisk 1.2.5 , and what is happening is this: Sip user agent "A" calls a pstn "phone B" Sip User agent Activates MOH. "B" starts listening. "A" doesn't hangup and just Disconnect Sipoftphone XLITE (exit) "B" stills listenning Music on Hold and "A" has left Asterisk, who hangs the call? only when B hangs...

hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/s at skype-web-callback-dial-263to263-1775,1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]:

Is there anyway to have multiple VOIP phones (from inside NAT firewalls and not) connect to my single * server? What do I need? I could put my * server on the outside of the my firewall but I'd rather not. Does the SIP Express server help at all? I can get phones to connect but I dont get any voice. I'm assuming it's NATn issues.

Hi, I'm trying to make native music on hold work on my Asterisk 1.2.9.1 server with a Sangoma PRI card. If I use a IAX phone connected to the PBX, I hear the music, but if I make a call from outside I hear nothing even if Asterisk console says music has started... it seems something related to zapata.conf but I cannot understand what's wrong. I also put musiconhold=native for every

Hey all; I am experiencing an issue with music on Hold. I am running asterisk version 1.4.22, and have a default script set up in two places for MoH playback. For internal devices to my network that are SIP peering with asterisk, they simply dial 123 and hear the MoH music immediately. I'm using the default setup, where it just plays the wave files in the /var/lib/asterisk/moh directory. I

Hello everyone, I am trying to activate Music On Hold using DB on Asterisk 13. It works fine but in order to use new Music On hold definitions I have to reload the moh module. - The following is my configuration in extconfig.conf - I added the following line: musiconhold.conf => mysql,asterisk,bit_ast_config - The following is the table in the database: mysql> select * from

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:

We're setting up asterisk at the office (really doing some testing right now) and it is going to be hosted on a dual G5 XServe running OS X. We're an apple certified solutions provider, etc. so we want to build all our stuff on apple hardware and software. Anyway, the last sticking point is moh and meetme. Is there any solution to get moh and meetme working on OS X? Meetme

Im completly new to the whole PBX thing. I have a toshiba unit now and we're moving our office in the next few months. I want to use asterisk but would like to test it out first. Does it support a basic analog phone line like the one in my house? Is that FXS? Are there any FAQs I should read to learn more? Thanks for the reply! -------------- next part -------------- An HTML attachment was

OK. So I did a test last night. All of asterisk's threads where using 0.0% CPU. I made 1 call to our call queue. CPU jumped to average of 9% and stayed around that for the 2 minutes I was in the queue just listening to music on hold. MOH is in MP3 format and I'm using format_mp3. Phone was linksys PAP2-NA using G729. Can I reasonably assume that the 9% was decoding the MP3, then

Hello: I have read that music on hold requires a timing source (which I never had to worry about previously since the server had zaptel hardware in it)...now I'm configuring a server in a colo which has no zaptel hardware. If I use the dialplan to run MusicOnHold(), I do get the music upon dialling that extension, but if I try to use the hold button on either a 7960 or X-Lite I get

Hello, Using asterisk-1.2.1 I am trying to convert my music-on-hold files from .wav to alaw: % sox moh.wav -r 8000 -c 1 moh.al resample -ql The file sounds fine when listened with: % sox mox.al -t ossdsp /dev/dsp But when listened through asterisk with an alaw SIP phone the sound is clicky and too fast. Did I forget something in my conversion command? -- ldm@apartia.fr

I am having a problem getting music on hold working one of my servers. I have had this working on a PII 400 just fine but decided to upgrade my Asterisk server to a PIV 1.5ghz. I have installed mpg123 which seems to be working fine but when I start *, I get the following error message at the CLI prompt when I start *: WARNING[81931]: File chan_oss.c, Line 232 (sound_thread): Read error on