search for: __transfer_context

Hi, I am in a weird situation where a variable seemed ignored, but not always. That variable is __TRANSFER_CONTEXT. Basically, I have a phone registered with asterisk. It's context is "internal". Outgoing calls go through that context (all good). When I get an incoming call which I want transferred, I don't want it to go through the context "internal" but through "inte...

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext 101 he gets connected to 37129238254 and after pressing # and 102# gets connected to ext SIP/12345. With a2billing (called via AGI), after pressing # and 102# the call gets...

Hi I'm using a macro to dial in a AEL dialplan. The problem is the macro do not set the field CDR(dst), showing only ~~s~~. I tried various configurations, but without solutions. This is the macro: macro dial-out(destno,dialstring,route_descr,interno) { __TRANSFER_CONTEXT=ipbx; if(${interno} = 1) { Set(__PICKUPMARK=${destno}); if(${ODBC_verify_user(${CALLERID(num)})} > 0) { t = tT; } else { t = t; } } else { t = T; } Dial(${dialstring}/${destno},30,${t}); return; } Thank's. Att, *Rafael dos Santos Saraiva* <http://br.linkedin.com/pub/rafael-saraiva/52/...

I'm trying to address a problem with users transferring to invalid destinations. In my sip peer I'm setting both __FORWARD_CONTEXT and __TRANSFER_CONTEXT to a context with a extension defined below to set some CDR variables before the call is transferred. [customer-forward] exten => _X.,1,Progress() exten => _X.,n,Gosub(do-billing,s,1${EXTEN})) exten => _X,n,Goto(customer-internal,${EXTEN},1) Now if my user Dials an invalid extension, Say...

hello,everyone! I was setting up ChanSpy in an Asterisk dialplan today and it just wasn't working. Here is the snippet: extensions.conf: [test] exten=>3001, 1, Set(__TRANSFER_CONTEXT=tranfer) exten=>3001, 2, Dial(SIP/3001,10,tr) exten=>2002, 1, Dial(SIP/2002,10,t) exten=>2002, 2, Hangup() exten=>2001, 1, Dial(SIP/2001,10,t) exten=>2001, 2, Hangup() exten=>1234, 1, ChanSpy(SIP,b) [tranfer] exten=> 2002...

asterisk-usershello,everyone! I was setting up ChanSpy in an Asterisk dialplan today and it just wasn't working. Here is the snippet: extensions.conf: [test] exten=>3001, 1, Set(__TRANSFER_CONTEXT=tranfer) exten=>3001, 2, Dial(SIP/3001,10,tr) exten=>2002, 1, Dial(SIP/2002,10,t) exten=>2002, 2, Hangup() exten=>2001, 1, Dial(SIP/2001,10,t) exten=>2001, 2, Hangup() exten=>1234, 1, ChanSpy(SIP,b) [tranfer] exten=> 200...

Hi, I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer. My extensions.conf: [globals] __TRANSFER_CONTEXT = transfer [common] exten => 123,1,Playback(demo-congrats) exten => 123,n,Hangup() exten => _0X.,1,Dial(SIP/${EXTEN}@PSTN-GW,60) exten => _0X.,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() [transfer] exten => 123,1,Goto(common,${EXTEN},1)...

Hi I'm using a macro to dial in a AEL dialplan. The problem is the macro do not set the field CDR(dst), showing only ~~s~~. I tried various configurations, but without solutions. This is the macro: macro dial-out(destno,dialstring,route_descr,interno) { __TRANSFER_CONTEXT=ipbx; if(${interno} = 1) { Set(__PICKUPMARK=${destno}); if(${ODBC_verify_user(${CALLERID(num)})} > 0) { t = tT; } else { t = t; } } else { t = T; } Dial(${dialstring}/${destno},30,${t}); return; } Thank's. Att, *Rafael dos Santos Saraiva* <http://br.linkedin.com/pub/rafael-saraiva/52/aa...

I'm confused. What does Asterisk do when it gets a 302 with a new number to forward to? Is there anything I have to do in the dialplan to make this work? I can't find any clear documentation on this issue.

...xited non-zero on 'SIP/voip.myserver.tld-081bfe98' -- Executing [860809864 at default:1] NoOp("SIP/voip.myserver.tld-081bfe98", "Receiving call to 860809864") in new stack -- Executing [860809864 at default:2] Set("SIP/voip.myserver.tld-081bfe98", "__TRANSFER_CONTEXT=default") in new stack -- Executing [860809864 at default:3] AGI("SIP/voip.myserver.tld-081bfe98", "/opt/itworks.php") in new stack -- Launched AGI Script /opt/itworks.php == Spawn extension (default, 860809864, 3) exited non-zero on 'SIP/voip.myserver.tld-081...

...an incoming call is handled - on 1.8 the below code recorded the entire conversation flawlessly, including the screening conversation of reception -> manager [inc] exten=>_[123]xxx,1,Macro(VCRECORD,${MACRO_CONTEXT}EXT${CALLERID(num)}ACC${CD R(accountcode)},${ARG2}) exten=>_[123]xxx,n,Set(__TRANSFER_CONTEXT=call-redirect) exten=>_[123]xxx,n(checkacc),NoOp(MY Account code is ${CDR(accountcode)}) ;exten=>_[123]xxx,n(checkacc),NoOp(OTHER Account code is ${CDR(accountcode)}) exten=>_[123]xxx,n(dodial),Dial(Sip/${EXTEN},120,tTg) exten=>_[123]xxx,n,NoOp(Dialstatus: ${DIALSTATUS}) exten=>_[123...

Hi, I want to check the status of a blind transfer (only sip endpoint) between various phones. Transfer is working perfectly, using ## from features.conf or using transfer key from phone, here SNOM320. My problem is that if party to transfer to is busy, the transfer fail and the call is ended. What I want to do is to return the call to the party who originate the transfer. I checked

Greetings, folks. I'm having a problem with blind transfers. It seems that, despite not having the T flag set, callers are able to use the blind transfer option. Scenario is this: - Asterisk 1.2.14 - Caller calls into our call center on one of our many phone numbers. - Call gets placed into queue. - Operator answers call. - Caller is able to hit our blind xfer key sequence (#0) and dial

I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more