Hi Mike, I've tried updating my CDR's via the h exten but with no success. I've tried with both endbeforehexten=no and endbeforehexten=yes (in cdr.conf) but the value refused to appear in my CDR (even though I see the Set() application being executed in the console under the h exten). Thank you for your suggestion though...
Any other thoughts are welcome. Kind Regards, Harel Cohen -----Original Message----- Date: Mon, 12 Dec 2011 13:41:31 -0700 From: Mike Diehl <[email protected]> Subject: Re: [asterisk-users] Populate CDR issues To: [email protected] Cc: Harel Cohen <[email protected]> Message-ID: <[email protected]> Content-Type: Text/Plain; charset="iso-8859-1" On Monday 12 December 2011 4:28:17 am Harel Cohen wrote: > Danny, > > Why would you think this is a "circumvent"? I'm using a nice feature > of 1.8 where I can create any CDR field I like and populate it by > using the > CDR(<fieldname>) function. While all other fields that I created are > populated properly (however before the 'dial' commences) it seems like > at this point of the dial plan the CDR is closed for editing even > though I configured endbeforehexten=no in my cdr.conf. I agree, this is a perfectly valid use of the CDR. I do the same thing, btw. I think what you are seeing is that when your call starts, Asterisk creates a record, either in memory, or in a db transaction. When the call is torn down, the record is updated and committed to the db. The down-shot is that any changes you make to the db record get clobbered by this last update. I ended up making some of my updates in the hang-up phase via the "h" extension. See if that will do what you need. -- Take care and have fun, Mike Diehl. ------------------------------ Date: Thu, 1 Dec 2011 12:57:56 +0100 From: Harel Cohen <[email protected]> Subject: [asterisk-users] Populate CDR issues To: "[email protected]" <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset="us-ascii" Hello list, I'm trying to populate my CDR logs with values which are available after the call has started (e.g. signalling IP of remote user, media IP, codec etc.). While CHANNEL function give me all I need for the incoming leg (leg A), I can't get the relevant values for the outgoing channel. I've tried using the option 'U' with my dial command (execute subroutine for called channel after called channel answered but before the call is bridged). While this throws the correct information to the console it does not populate the CDRs accordingly. Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC and the table therein contains the relevant fields. This is the console with 'very-verbose' output for the 'Dial' application where office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP 192.168.20.226. My comments added prefixed by ** and on separate line: ** channel here is source channel: SIP/office_Admin2-00000015 [Dec 1 12:14:31] -- Executing [316@InternalDP:5] Dial("SIP/office_Admin2-00000015", "SIP/office_ServerRoom,,FgU(jump2SetVar)") in new stack [Dec 1 12:14:31] == Using UDPTL CoS mark 5 [Dec 1 12:14:31] == Using SIP RTP CoS mark 5 [Dec 1 12:14:31] -- Called SIP/office_ServerRoom [Dec 1 12:14:31] -- SIP/office_ServerRoom-00000016 is ringing [Dec 1 12:14:31] -- SIP/office_ServerRoom-00000016 is ringing [Dec 1 12:14:33] -- SIP/office_ServerRoom-00000016 answered SIP/office_Admin2-00000015 ** from here the channel is the destination channel: SIP/office_ServerRoom-00000016 [Dec 1 12:14:33] -- Executing [s@jump2SetVar:1] Gosub("SIP/office_ServerRoom-00000016", "SetVar,postdial,1") in new stack ** This is how I obtain channel information: ** exten => postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peername)},port)}) ** same => n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)}) ** same => n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)}) [Dec 1 12:14:33] -- Executing [postdial@SetVar:1] Set("SIP/office_ServerRoom-00000016", "CDR(chanoutsigip)=192.168.20.226:5065") in new stack [Dec 1 12:14:33] -- Executing [postdial@SetVar:2] Set("SIP/office_ServerRoom-00000016", "CDR(chanoutmediaip)=192.168.20.226:23008") in new stack [Dec 1 12:14:33] -- Executing [postdial@SetVar:3] Set("SIP/office_ServerRoom-00000016", "CDR(chanoutcodec)=g729") in new stack [Dec 1 12:14:33] -- Executing [postdial@SetVar:4] Goto("SIP/office_ServerRoom-00000016", "endsub,1") in new stack [Dec 1 12:14:33] -- Goto (SetVar,endsub,1) [Dec 1 12:14:33] -- Executing [endsub@SetVar:1] Return("SIP/office_ServerRoom-00000016", "") in new stack [Dec 1 12:14:33] -- Executing [s@jump2SetVar:2] Return("SIP/office_ServerRoom-00000016", "") in new stack [Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1] NoOp("SIP/office_ServerRoom-00000016", "") in new stack [Dec 1 12:14:33] -- Auto fallthrough, channel 'SIP/office_ServerRoom-00000016' status is 'UNKNOWN' [Dec 1 12:14:33] -- Remotely bridging SIP/office_Admin2-00000015 and SIP/office_ServerRoom-00000016 When call is terminated the relevant fields in the database for CDR(chanoutsigip), CDR(chanoutmediaip) and CDR(chanoutcodec) are populated with their default values (typically blank or '-----') and NOT with the values above. Am I doing something wrong or is there a different way to populate CDR's with info from called channel (leg B)? Thank you for your replies... Harel -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
