similar to: TRANSFER_CONTEXT ignored?

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

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

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. 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 =>

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

Hello! Most are probably bored seeing another letter about this, but I've put in a fair amount work on a spec for rewriting the CDR system in Asterisk, and I have some questions: First, please look at what I've written so far: svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs and look at the file "CDRfix2.rfc.txt" in the RFCs dir. The spec SIGNIFICANTLY alters the way

Hi all, There's been a few posts looking for telemarketer torture scripts so I figured that I would write one using a SQLite db. Handling an incoming call that is flagged from the database is pretty simple. My problem is that I would like the callee on an established channel to be able to redirect the caller to a specific context where my AGI is called and handles the call by first

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

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

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

Hello, We are in the process of splitting our phone system into two separate logical systems for our two departments. One of the goals of this switch is to restrict members of one department from transferring calls to the other, but not restrict them from calling that department themselves. So what I need to know is how to detect whether a call from a member of that department is a transfer or

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

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

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

Hi, I've thoroughly searched the mailing list for a concrete example but haven't found one, thats why I end up here with my question. Theres has been some talk about IAX Transfers, but quite theoretically. Lets say there is a setup of several asterisk-boxes connected in a local Ethernet. Asterisk A establishes actively a connection to a SIP client B. When B picks up, the connection

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

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

On Thu, March 5, 2015 09:56, Ruben R?gels wrote: > > Hi again, > > I'm glad to hear that I provided a somehow useful answer. > > Unfortunatelly, I don't know these details. > If you wasn't lucky consulting the snom docs, maybe the snom support > can be helpful with information about the exact implementation > details. > > You also could use "sip

Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL queries). Anything I can do? Mick -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

Hello, I'm trying to find more information about this Remote Attended Transfers, as is explained in https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers for Asterisk 12 using pjsip stack Was Remote Attended Transfer implemented in previous versions of Asterisk (versions without PJSIP, Asterisk 11 and previous)? Where can I find configuration examples to do it work