similar to: turn the ring tone OFF during dialing

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

Hi All, I might of got my wires crossed here, but I'm looking for a way to disable musiconhold for individual users. I had thought that putting the sip.conf entry to: [690] type=friend context=from-sip secret=***** qualify=yes host=dynamic canreinvite=no nat=yes mailbox=2090 callerid=2090 musiconhold=silent and then putting an entry in musiconhold.conf like: [silent]

i've tried to set dial = 0/1500 in the indications.conf but still getting a dial tone. is this a bug with *? or did i do something wrong. thank you, patrick

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 >

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten

Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other

I'm testing GotoIf($["${CALLERID(num) but I'm missing something as it is not working: [office-open] exten => s,1,Wait(1) exten => s,2,Answer() ; for Caller ID is 471-5665, always signal congestion: exten => s,3,GotoIf($["${CALLERID(num)}" = "4715665"]?4:6) exten => s,4,Playtones(congestion) exten => s,5,Congestion(5) exten =>

Hi everybody, I don't know how to do this, I redirect a call and then dial someone, but I dont want the ringing tone to be listened while the dialing part is waiting for the dialed part to pick up the phone. exten => 555,2,Dial(${STRING4},30) I have tried when the option 'm' , but I don?t want the default music on hold to be listened neither. I want nothing (silence) to be heard

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with

Hiya, I''ve recently been looking at removing the dependency in our CRM/CMS on MSIE XML Databinding so that it''ll run properly in Firefox.. The solution I''ve created uses the prototype.js lib so I thought I might share my efforts here if that''s ok? It''s still _very_ early days but there''s the beginnings of a writeup here:

Hi There, I have an asterisk an a Digium 4 Port E1 Card On E1 Port No. 1 I have the Telekom PRI On E1 Port No. 2 I have an Alcatel PBX that cannot be changed So I have setup my asterisk between Alcatel and Telekom In extension.conf i configured: [telekom] exten => _9149.,1,Dial,ZAP/g2/${EXTEN}; exten => _9149.,2,Hangup This works great, all incoming calls are directly routed to alcatel

Hi, I have a problem with TDM2400P card. The card is detected ok, I can make a call but only with pulse dialing (not tone dialing) without hear sounds from the headset. When I receive a call, I can to establish a communication, but without hear sounds from the headset. When I dial any phone key, I can hear dtmf tone. I'm using Elastix 1.5.2. These are my configuration files:

Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. For example: [test] exten => 123,1,PlayTones(ring) exten => 123,n,Wait(5) exten => 123,n,Playback(demo-congrats) exten => 123,n,Hangup() exten => 321,1,Dail(LOCAL/123 at test/n,60,r) When I now dial with a SIP phone - 123 I can hear nice

Hello, I'm trying to do something really easy : transfer a PSTN call to a H323 client. This works great. Now I'm trying to use the SetMusicOnHold function. I din't find any doc about it, I've just seen some mails in the list archive, but it still doesn't work. That's my extension.conf : [incoming] exten => s,1,SetMusicOnHold,default exten =>

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

Olly, Thanks again very much for helping me improve my understanding of Xapian and Omega. Thanks especially for pointing out that my idea of trying to generate a snippet from stemmed text lacking capitalization and punctuation would probably not produce a user-friendly result. But I'm still doubtful that expanding the sample size could be the right way to obtain excerpts from the document

Question: I'm using a .call file to make some test calls. The call file works great. When I try the same thing with the manager 'originate' action I get something weird - the originate action looks for the 's' extension in my context, regardless of what I supply as the 'extension' argument. The .call file does what I expect - it finds exten _9.,1,Noop(Looks good).

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

Hello all. I am just getting back into Asterisk and I am setting up my Linksys SPA3102. I have incoming calls working fine, as is the phone plugged into the unit. My problem is I cannot get the SPA3102 to dial a phone number automatically. I can call the extention of the PSTN and I get a second dialtone, and I can then manually dial. I'd like to be able to have Asterisk pass the