search for: my_context

I am using the AMI to dispatch (2) calls to Local/my_priority at my_context where: [my_context] exten => my_priority,1,Answer() exten => my_priority,n,Dial(${LOCAL_DIAL}) and LOCAL_DIAL has the actual phone number to dial. The first call goes through just fine and I see DAHDI/1/XXXX being called. The second call I see DAHDI/2/XXXX and a message about everyone is...

Hi, I'm Ryo Ota. I have two questions about `llvm::LLVMContext`. Q1) What is the difference between the following (A)`my_context` and (B)`global_context`? Do I have to create a LLVMContext by myself? Or use `getGlobalContext()`? Could you tell me what situation needs a LLVMContext which is created by myself such as (A)? (A) { llvm::LLVMContext my_context; // codes using only my_context (get Int32ty() or StructType::create o...

...11.5.1. As far as I understood, the following configuration allows a sip client only to receive calls (type=peer) but not to place calls (http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place calls though with this config? sip.conf ... [thorsten] type=peer host=dynamic context=my_context nat=force_rport,comedia secret=... dtmfmode=rfc2833 disallow=all allow=g722 allow=g729 allow=g729 ... extensions.conf ... [my_context] exten => _X.,1,Dial(DAHDI/g1/${EXTEN},60) ... Of course: when removing a valid context the client can not place the call. But I thought this behaviour can be...

I've isolated this problem the furthest that I can, and I'm now convinced this is a bug in asterisk. I have a context in extensions.conf like so: [my_context] exten => _X.,1,AGI(my_agi|${EXTEN}|${CHANNEL}) exten => _X.,2,GOTO(my_other_context|${EXTEN}|1) exten => h,1,DeadAGI(my_agi_cleanup) For the purposes of this scenario, my_agi simply will try to HANGUP the channel to avoid the call going to priority 2 and instead go to my_agi_cleanup....

Hi all, I've tried to sen calls to asterisk from different soft switch. I want to define ip authentication(not register) to an extension for make call through asterisk. Is there any way to make call from asterisk without register. Only ip authentication. I tried too many different configurations but it hasn't worked. This is my sip.conf --sip.conf [3333] host=x.x.x.x qualify=yes