[asterisk-gui] CallerID not working on PSTN with macro-trunk-dial-failover

bkruse bkruse at digium.com
Wed Oct 15 17:03:35 CDT 2008


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 [4220914 at DLPN_Open:1] Macro("SIP/6001-08cff1f0", 
> "trunkdial-failover-0.3|Zap/g1/4220914|Zap/g2/4220914|trunk_1|trunk_2") 
> in new stack
>     -- Executing [s at macro-trunkdial-failover-0.3:1] 
> Set("SIP/6001-08cff1f0", "CALLERID(num)=") in new stack
>     -- Executing [s at macro-trunkdial-failover-0.3:2] 
> GotoIf("SIP/6001-08cff1f0", "0?1-dial|1") in new stack
>     -- Executing [s at macro-trunkdial-failover-0.3:3] 
> Set("SIP/6001-08cff1f0", "CALLERID(all)=") in new stack
>     -- Executing [s at macro-trunkdial-failover-0.3:4] 
> Goto("SIP/6001-08cff1f0", "1-dial|1") in new stack
>     -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
>     -- Executing [1-dial at macro-trunkdial-failover-0.3:1] 
> 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 <bkruse at digium.com 
> <mailto:bkruse at digium.com>> 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




More information about the asterisk-gui mailing list