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The wiki says this about the MusicOnHold command: "Plays hold music specified by class. If omitted, the default music source for the channel will be used." http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold How do I set the default music on hold class for the SIP channel ? I tried adding musiconhold=test to my sip.conf. musiconhold.conf looks like this: [classes] default => quietmp3:/var/lib/asterisk/mohmp3 loud => mp3:/var/lib/asterisk/mohmp3 random => quietmp3:/var/l...

Is it possible to get an internal dial tone when I call to my asterisk and enter password? I would like to call my line enter extension - password - and get internal dial tone. once I'm in I would like to dial based on what context permits, mostly long distance calls VOIP. I can not preset the extension to certain number as I don't know what number I will be dialing. -- #Joseph

I'm looking into solutions for providing a live stream of an event in Belgium [1] - for example, as follows: * Event --> mobile phone --> software answering machine --> Internet server * Event --> mobile phone --> VOIP --> Internet server The live stream should be available in a format so that people can listen to it using XMMS or similar software. Comments? Would

...ltone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to"... Some buddy of mine told me im looking for a function called "flash" Only thing Im able to find is: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash Im unsure how to use it now.. Let's say if I forward a call with asterisk as following: exten => 2,1,Dial(capi/720****:07812345*,18) How would I use the flash command to transfer that call above to 078 12345* ? I have no problem transferring a call, but when Im doing this with the...

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

...ng CentOS-5 a while ago and so far it has worked out great. I crosspost to r-sig-hpc and welcome others to comment if there are potential downsides to this solution. George > Michael Spiegel michael.m.spiegel at gmail.com > <mailto:r-devel%40r-project.org?Subject=Re%3A%20%5BRd%5D%20R%20CMD%20INSTALL%20configure.args%20and%20CC%20customization&In-Reply-To=%3CCANwu5-rGUYrC73vRo04GJCdW_ZQerycbVU4K_2Dun1ytqNXQ7g%40mail.gmail.com%3E> > /Fri Sep 9 20:40:42 CEST 2011/ > I am running into the following issue that has been previously > reported on the R-devel mailing list. Th...

Hi, How do you transfer (using say blind transfer) a call to an extension that is currently busy on another call? You don't want the call to be transferred to voicemail, it must stay in 'hold' until the extension becomes available, and then immediately ring that phone. Thanks, Thomas

Finally got everything up and run with the help of Manny Wise last night. So I am setting up my digital assistant and getting down to the task I need this box to perform the most. I need to have a custom app that I can call that will take me pressing 2 at the menu and have it transfer the call to a offsite phone number utilizing my Zap Trunk. I'm sure someone has done this already. Anyone want

I have Asterisk setup in a SOHO environment. I have 2 X100P cards at Zap/1 and Zap/2. I have 1 TDM400P card with Zap/3 - Zap/5. I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2. My problem is when I'm in an active call to the outside thru Zap/1 or Zap/2, I can't pickup the incoming callwaiting call. I can see the

Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone's post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler -------------- next

I have a client that has an old Merlin system. They would like to move to an Asterisk based system, however, with their existing system each phone is capable of displaying who is on the phone within there office. This is done by lighting a red light for each line(extension) that is in use. Has anyone been able to neatly create this feature? Perhaps an XML application can be written for the Cisco

hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar

Any one know of software that allows 2-way radios as VoIP(SIP) clients, besides dingotel's usb & mic cable trick ? http://www.dingotel.com/2way/requirements2way.asp They might be ok if the SIP client was not hardcode to their own SIP proxy Has anyone tried any hacks to get the 2-way radio SIP client to regsiter to a * box. hmm chan_frsgmfrs anyone? using the usb/mic cable under linux :)

Am a little confused on how I can design a dial plan like this: extension 700 should prompt for a PIN and password lookup mysql to see if this is correct if correct, log the time / date in mysql announce "logged" and hangup any ideas? Thanks a bunch --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! -------------- next part

Hi everyone, I think asterisk is really great, but since I started sending sms using * I've some troubles with it! I setup everything as it is described at voip-info.org (http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Sms) and it really works: I can send SMS - as long as they are shorter than 32 characters. btw I'm living in Germany and therefore I send my SMS via T-Com (they use the same protocol BT uses). Does anyone know what's going wrong? Bye, Daniel the relavant part of my extensions: ;;;;;...

Hello, I'd like to be able to see if a channel is use and handle the call differently if it is. The best I can find is the command ChanIsAvail(). The problem is, I have an snom200 phone which does call waiting, so even if it is engaged in a call, a second channel is still available on it. I would like to be able to differentiate between these two cases: no calls engages, or calls

I can see how you log an agent in, but how to log one out? For login I think I'll have our people dial something like _8900XXXX, then use $(EXTEN:4} to get the callback extension automatically I can't find any better way at the moment, all my outgoing CID info needs to point to our main trunk line, but I'm not sure I can make that happen any other way than what I'm doing now

I am trying to figure out the correct way to send my E.164 ID with chan_h323 and or chan_oh323 as my H323 provider requires this in the format of 'account-pin'. With chan_oh323 I have been able to register with the gatekeeper and can recieve incomming calls, but outgoing calls do not work. With chan_h323, I can call H323 clients (netmeeting, ATAs etc) but cannot place a call through my

'http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA' This should help. -----Original Message----- From: Michael Graves [mailto:mgraves@mstvp.com] Sent: Wednesday, January 26, 2005 9:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] dialplan logic for conditional DISA on incomming 800 number...

http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetGroup -Matthew ----- Original Message ----- From: "Stefan Gofferje" <stefan@gofferje.homelinux.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, January 31, 2005 6:43 PM Subject: [Asterisk-Users...