Jonas Kellens
2017-May-29 07:17 UTC
[asterisk-users] Best way to know a call is being transfered
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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170529/76c60956/attachment.html>
Jonathan H
2017-May-29 08:16 UTC
[asterisk-users] Best way to know a call is being transfered
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.kellens at 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
Jonas Kellens
2017-May-29 09:06 UTC
[asterisk-users] Best way to know a call is being transfered
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.kellens at 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