No problem Marvin, glad I could help :) -bk
Marvin Whitfield wrote: > I tried your recommendation and it WORKED! For those who might have a > similar problem, you have to set a Caller ID value > 6 digits and your > extension CID info will pass to CDR. Only problem I have with this is > that setting an outbound CID number for PSTN really has no effect so > the number that you set is just so everything functions correctly and > has no real effect. > > Much thanks bk > > -- > Marvin > > On Wed, Oct 15, 2008 at 5:03 PM, bkruse <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>> wrote: > > > Your callerID is just not setup properly. > > Set a CallerID for the trunk you are dialing out, and it will use > that one > > -bk > > Marvin Whitfield wrote: > > So it seems like I stated the problem incorrectly...What seems to > > really be happening is that for outbound calls on Zap channels there > > is no CID info. > > > > My cdr_custom map is: > > Master.csv => > > > > "${CDR(src)}","${CDR(dst)}","${CDR(clid)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(start)}","${CDR(end)}","${CDR(duration)}","${CDR(disposition)}" > > > > Sample output: > > "","4220914","","SIP/6001-08cff1f0","Zap/1-1","Dial","2009-10-14 > > 16:30:26","2009-10-14 16:30:34","8","ANSWERED" > > > > CLI output: > > > > Executing [EMAIL PROTECTED]:1] Macro("SIP/6001-08cff1f0", > > > "trunkdial-failover-0.3|Zap/g1/4220914|Zap/g2/4220914|trunk_1|trunk_2") > > in new stack > > -- Executing [EMAIL PROTECTED]:1] > > Set("SIP/6001-08cff1f0", "CALLERID(num)=") in new stack > > -- Executing [EMAIL PROTECTED]:2] > > GotoIf("SIP/6001-08cff1f0", "0?1-dial|1") in new stack > > -- Executing [EMAIL PROTECTED]:3] > > Set("SIP/6001-08cff1f0", "CALLERID(all)=") in new stack > > -- Executing [EMAIL PROTECTED]:4] > > Goto("SIP/6001-08cff1f0", "1-dial|1") in new stack > > -- Goto (macro-trunkdial-failover-0.3,1-dial,1) > > -- Executing [EMAIL PROTECTED]:1] > > Dial("SIP/6001-08cff1f0", "Zap/g1/4220914") in new stack > > -- Called g1/4220914 > > -- Zap/1-1 answered SIP/6001-08cff1f0 > > -- Hungup 'Zap/1-1' > > > > So it seems CALLERID(all) is being Set to nothing on line 2 > (line 3 in > > the macro). I haven't assigned callerid to any of my extension as I > > expected the extension info to be passed (as happens when I comment > > out line 3). > > > > Hope this helps > > > > - Marvin > > > > > > On Wed, Oct 15, 2008 at 3:48 PM, bkruse <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]> > > <mailto:[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>> wrote: > > > > > > Hey Marvin, sorry for the delayed response. > > > > My guess is that you do not have all the callerid fields > properly set. > > > > Can you tell me what your: > > > > User's callerID calling is > > > > Trunk CallerID > > > > and Global CallerID > > > > The order in which we try callerid's is User -> (if it is > invalid > > then) > > -> Trunk (if it is invalid then) -> Global. > > > > Also, if you paste the output of the Asterisk CLI when the > call is > > made, > > we can see what is going on :) > > > > Thanks, > > > > -bk > > > > Marvin Whitfield wrote: > > > I have been having some trouble with the new macro-trunk-dial > > > regarding callerID. Before bk posted the new CDR page I > made on > > for my > > > self that gets values from the Master.csv (cdr-custom). I > > noticed that > > > for incoming calls on PSTN channels no callerID info is passed > > to CDR. > > > In both cdr-custom and normal cdr there was no callerID > information. > > > There was still all the other information. I took a look > at the > > macro > > > and realized that the Set() function must be the culprit and > > commented > > > out the third line for testing and it worked. I am hoping > that there > > > is a better solution since the GLOBAL CID will be lost for > digital > > > channels but I don't see why it can't work for PSTN since > you can't > > > send outbound CID info anyway. Any ideas? > > > > > > -- > > > Thanks, > > > Marvin > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > > > > > asterisk-gui mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-gui > > > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > > > asterisk-gui mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-gui > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-gui mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-gui > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-gui mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-gui > > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-gui mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-gui _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui
