Unfortunately, the transfer AMI events were introduced just in Asterisk13. But, you can set the __TRANSFER_CONTEXT variable and probably the __GOTO_ON_BLINDXFR (this one I never used) to control the transfer in your own way.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables Regards, Marcelo H. Terres <mhter...@gmail.com> IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 29 May 2017 at 10:06, Jonas Kellens <jonas.kell...@telenet.be> wrote: > Hello > > thank you for your answer. > > However this does not help me to know when a call is being transfered. > > My question is simple : if A calls B, and then B tranfers (unattened or > attended) the call to C, how can I know this happens ?? I see it happening > on the CLI, but how can I "catch" this, for example in the dialplan logic ? > Or through AMI perhaps ? > > > > Kind regards. > > J. > > > > Op 29-05-17 om 10:16 schreef Jonathan H: > >> Well, once you've upgraded to a version of Asterisk which didn't >> become "EOL - DO NOT USE - NO FIXES" (!) almost 2 years ago, then you >> might be able use logging which was introduced 5 years ago in Asterisk >> 11. Although the "transfers" section in the info below says it "can be >> a little tricky...". Read on! >> >> https://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging >> >> ------------------------------------ >> >> Call ID Logging (which has nothing to do with caller ID) is a new >> feature of Asterisk 11 intended to help administrators and support >> givers to more quickly understand problems that occur during the >> course of calls. Channels are now bound to call identifiers which can >> be shared among a number of channels, threads, and other consumers. >> >> Transfers >> >> Transfers can be a little tricky to follow with the call ID logging >> feature. As a general rule, an attended transfer will always result in >> a new call ID being made because a separate call must occur between >> the party that initiates the transfer and whatever extension is going >> to receive it. Once the attended transfer is completed, the channel >> that was transferred will use the Call ID created when the transferrer >> called the recipient. >> >> Blind transfers are slightly more variable. If a SIP peer 'peer1' >> calls another SIP peer 'peer2' via the dial application and peer2 >> blind transfers peer1 elsewhere, the call ID will persist. If on the >> other hand, peer1 blind transfers peer2 at this point a new call ID >> will be created. When peer1 transfers peer2, peer2 has a new channel >> created which enters the PBX for the first time, so it creates a new >> call ID. When peer1 is transferred, it simply resumes running PBX, so >> the call is still considered the same call. By setting the debug level >> to 3 for the channel internal API (channel_internal_api.c), all call >> ID settings for every channel will be logged and this may be able to >> help when trying to keep track of calls through multiple transfers. >> >> >> On 29 May 2017 at 08:17, Jonas Kellens <jonas.kell...@telenet.be> wrote: >>> >>> Hello >>> >>> using Asterisk 1.8.32.3. >>> >>> What is the best way of knowing a call is being transfered (attended and >>> unattended) ? And also knowing whereto (sip user) the call is being >>> transfered and who is the transferer ? >>> >>> So I can log this information. >>> >>> >>> >>> Kind regards. >>> >>> J. >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Check out the new Asterisk community forum at: >>> https://community.asterisk.org/ >>> >>> New to Asterisk? Start here: >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users