On Tuesday 11 October 2011, Olivier wrote: > Hi, > > I'm facing a strange problem. > > My setup is: > Alice cellphone <--GSM--><--ISDN--> Asterisk <-- ISDN --><--GSM--> Bob > cellphone > > When Alice calls Asterisk which forwards the incoming call to Bob, > sometimes Bob sees Alice's number, sometimes he sees a default CallerID > (which happens to match the dialed number and the ANI). > For various reasons, Bob really needs to see Alice's number when Alice is > calling. > > When I compare one "successful" (ie presented with Alice ID) calls with one > "unsuccessful" (with debug and verbose levels respectively set to 0 and 3), > I can't see any difference between both calls within Asterisk logs : > every time Asterisk, receives Alice CallerID and set outgoing channel > CallerID with the same value. > (The only I could find, at the moment, to distinguish a successful call is > to call Bob and ask him to tell what happened). > > If that matters, let me add this: > - each incoming call is forwarded with a simple Answer(), Dial() sequence, > - when I'm presenting an outgoing with too many digits, the call is > presented with a default CallerID. > > My understanding is "as each network used is purely digital, you can't > loose CallerID". > Is this roughly correct ? > In which direction, shall I dig ?
Most telcos won't let you present a caller ID number that doesn't belong to you; so it's possible that the number you are presenting to Bob is being munged on the way to his mobile. Otherwise, anybody with the right equipment would be able to pretend to be anybody else, and caller ID would be all but useless. Anyway, what you really need to do is separate the two legs of the call, to see whether the number is changing between Alice and your Asterisk or between your Asterisk and Bob. So put a Verbose() or NoOp() in your dialplan to see what caller ID Alice is sending, and get her to call you several times. Then create a context to call Bob (presenting Alice's number) from a SIP phone, and call him several times. -- AJS Answers come *after* questions. -- _____________________________________________________________________ -- 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