similar to: SIPPeersAction class file not found in the Asterisk-java.jar file

hello all, I would like to create an application where a file is played repeatedly in the background while two parties are having a conversation. Does anyone know of a way to achieve this? I have been looking into the ManagerAPI to redirect the call to a meetme room. Then I try to introduce a third party that will just play the file continuously, but with no luck. Is there a different/simpler

For an inbound call center with 4 T1s and 30-50 agents on you would do just fine with a single, one-processor machine. We have handled more than this on a single P4 server although we use astGUIclient instead of Asterisk queues, but the load is very similar. I would recommend a Sangoma Quad T1 card because they are about 30% more efficient than Digium T1 cards. When you say that you need to

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > Tested with X-Lite and it worked fiine. Is there some way to replace > "Anonymous" with a config parameter? > > Thanks for your kind help > > ---------------------------------------- > > From: murthy64 at hotmail.com > > To: asterisk-users at lists.digium.com >

On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:07:35 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > <snip> > >> Here

Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

Hi, I have a problem dialing a SIP phone which is logged in as different Astesrik machine from the one I am working with. I want to call a phone in Another astersik machine in , if it answers, calling a SiP phone registered in my ASterisk: My dialplan is: [mariaSIP] exten => _1.,1,Wait(1) exten => _1.,2,Dial(SIP/6021@192.168.0.51:5038,20) exten => _1.,3,HangUp() exten =>

On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > > > ________________________________ > > Date: Thu, 6 Aug 2015 12:55:28 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > > > > > On

Try Capitalizing Your Actions And The Parameters For Them: Action: Login Login: admin Secret: mypass Action: Originate Exten: 200 Context: stations Channel: SIP/agent07 Priority: 1 Callerid: James Bond Calling That should work. We use manager actions extensively in our applications and the managerAPI is sometimes finicky for capitalisation. and a capitalised first letter of the action and

With Asterisk-Java the proposed solution to connect to multiple Asterisk servers is to create multiple AsteriskManagerConnection obeject. Each ManagerConnection handles its own thread so there is no need for custom thread handing code. All you have to do is to make sure is the EventListener objects you pass to these connections synchronize access to shared data (if there are such accesses). I