search for: transfer_context

Hi, Can anyone suggest a way of doing the following in Asterisk 1.6.2 - I do not think it can be done trivially using TRANSFER_CONTEXT. What I want is for the TRANSFER_CONTEXT for all technologies to be the same as the initial context defined in the configuration of the device initiating the transfer. This is not as simple as it seems (unless I am missing something). For example: A call arrives on IAX/1234, and executes DIAL(SIP...

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...

...ures.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 variable like ${BLINDTRANSFER} ${TRANSFERED_BY} or ${TRANSFER_CONTEXT}, they are all empty. What did I miss? Thanks for any hints -- Daniel

...isk)--------->>>>(IAX|SIP??)>>>---------C(Asterisk) Asterisk A tries to transfer the SIP connection to Aseterisk C Will that lead to B---------(SIP)-----------C (???) How would this look like in the dialplan? Lets say the originated Call from A to B ends up in context [Transfer_Context] A: [Transfer_Context] transfer => ...... [Transfer_Context] extension => X,X,Transfer(Y) How fast would the transfer be if C picks up immediately, would B be able to recognise this? Thanks in advance, Knud

I am trying to setup a couple of virtual pbx's off of my one may asterisk box. So far I have been able to segment most everything via the Dial plan. My only question/problem has to do with the # Transfer function. I had set up # Transfers prior to segmenting the dial plan, and I cannot remember how I was able to specify which context to use when the user presses #. I haven't been able

...led and handles the call by first recording the callers CID into the database (if available) and then starts to turn the screws on the caller. Is it possible to conditionally override the blind transfer feature somehow to achieve this functionality? I've looked a Transfer() cmd and setting TRANSFER_CONTEXT, but they don't seem like what I need as they do not provide control during a call. Can this be implemented in the context, "context=>parkedcalls" specified in features.conf? Thanks, -- Warm Regards, Lee

I know this isn't an asterisk thing, but since the recommendation to get one came from here I figure lots of people out there have one. I read the docs, and it says that in order to do a blind transfer I should hit "flash", then dial "*__" then the number. Now, how on a normal phone do I dial "asterisk underscore underscore"? Can someone tell me how doing a

OK, it's been a while since I drank from the pool of wisdom hear on the list. After cracking my head against the wall for a few days trying to figure this out, I have decided to swallow my pride and take the drink. So, on to my question: I have some agents/operators setup in sip.conf which point to a context where I have just about disabled outbound calls (only specific numbers can be

...Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten => s,1,Set(TRANSFER_CONTEXT=transfer) exten => s,2,Set(FORWARD_CONTEXT=transfer) exten => s,3,Set(CIDNUMORIG=${CALLERID(num)}) ; save callerid-num exten => s,4,Set(CALLERID(num)=98765) ; dial out using msn 98765 exten => s,5,Dial(Zap/g1/${ARG1}|30|t) exten => h,1,Set(CALLERID(num)=${CIDNUMORIG}) restore caller...

Hello, while most of the physical phones have keys to handle attended and blind transfer, most soft phones have no support for it. Asterisk offers a "featuremap" to assign a key to blindxfer and atxfer and they work fine if the call is still in the same starting context, but if the call has moved in another context, then the new call will be started from such context with unpredictable

...ytrunk aors=mytrunk [mytrunk] type=identify endpoint=mytrunk match=89.1.23.217 extensions.conf [transferencia] exten => external_replaces,1,NoOp() same => n,Dial(PJSIP/${SIPREFERTOHDR}@89.1.23.217) [bucle-weasels] exten => _.,1,Answer exten => _.,n,Wait(1) exten => _.,n,Set(TRANSFER_CONTEXT=transferencia) exten => _.,n,Playback(tt-weasels) exten => _.,n,Goto(2) exten => _.,n,Hangup Thank you in advance for your help David Pinedo -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20...

Hi list! When I'm doing transfer, to what context/priority does that call goes? Can it be changed? Is it the same for blind_tr/att_tr/and for transfer that appears when phone replies with - 302 "Moved Temporarily"? The thing is that I'm trying to transfer incoming call from E1 interface back to E1 interface. Transfers will occur when user is going out and sets up all call

...t's too large and probably useless without lot of external configs. I can just tell you where to look in info, and if you don't have something working as expected - you're welcome to ask. So, please see http://www.voip-info.org/wiki/index.php?page=Asterisk+variables - there is variable TRANSFER_CONTEXT, you have to set before dialing from A to B. Also, you have to set some global variable that will have extension B (it's hard to acquire it later) Then you should define the context that will handle transfers from B to C. It should contain Dial() command to extension C. You need to have parame...

Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager

Hi! I have some problems understanding the concept of REFER in Asterisk 1.4.23. I have the following scenario: Incoming SIP call (incoming leg) from a SIP trunk into Asterisk (handled in context fromTrunk), forwarded to the SIP Client (outgoing leg). Now, the SIP Client sends a REFER request (unattended transfer) to another extension. This terminates the outgoing leg and the incoming leg

...arted music on hold, class 'default', on SIP/570-00000006 [Jun 4 22:30:31] VERBOSE[11301] file.c: -- <SIP/518-00000005> Playing 'pbx-transfer.gsm' (language 'nl') [Jun 4 22:30:32] WARNING[11301] features.c: Extension '53' does not exist in context 'transfer_context,570,1' [Jun 4 22:30:32] VERBOSE[11301] file.c: -- <SIP/518-00000005> Playing 'pbx-invalid.gsm' (language 'nl') [Jun 4 22:30:34] VERBOSE[11301] res_musiconhold.c: -- Stopped music on hold on SIP/570-00000006 Mind you, the entered extension is "531", s...

Is there a general recipe to avoid fraudulent calls under the following conditions? A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to call and then transfer to him), i.e. the Dial cmd for the internal context contains "Tt". Then an outside call would operate as a Local channel in an internal context after the first transfer. If the

...ot;MYCONTEXT=5000001") in new stack -- Executing Set("SIP/3013-3dfa", "CALLERID(name)=Cypress") in new stack -- Executing Set("SIP/3013-3dfa", "CALLERID(number)=2814494000") in new stack -- Executing Set("SIP/3013-3dfa", "TRANSFER_CONTEXT=cytel-internal") in new stack -- Executing GotoIf("SIP/3013-3dfa", "0?global-outgoing|17034439032|1") in new stack -- Executing Dial("SIP/3013-3dfa", "SIP/17034439032@22.22.22.22|60|WT") in new stack -- Called 17034439032@22.22.22.22...

Hi, Does anybody know if there is a way for a call goes back to transferer if unanswered ? Thanks Luis A P Barbosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070912/1e356013/attachment.htm

Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a