On 10-07-12 19:48, Warren Selby wrote:
On Tue, Jul 10, 2012 at 12:34 PM, Patrick Lists <[email protected] <mailto:[email protected]>> wrote:Thank you for your feedback Warren. I removed the outbound name but still get random numbers & "VOIP CALLER" on outbound calls. Googling I tried some more: SipAddHeader(P-Asserted-__Identity: <sip:19995551212@AST_BOX_FQDN>__) SipAddHeader(P-Asserted-__Identity: 19995551212) SipAddHeader(P-Preferred-__Identity: <sip:19995551212@AST_BOX_FQDN>__) SipAddHeader(P-Preferred-__Identity: 19995551212) But none of them work. So unless someone has the magic incantation howto make this work I'll open a ticket with flowroute. I use Flowroute. My outbound callerID is set as follows: [outgoing] exten => _X.,1,Verbose(Outound call from ${callidnum} to ${EXTEN} on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}) exten => _X.,n,Set(CALLERID(num)=${callidnum}) exten => _X.,n,Goto(outgoing-dial,${EXTEN},1) [outgoing-dial] exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@flowroute) exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@flowroute) ${callidnum} is a variable from my SIP peer (setvar=callidnum=7133437300). This always passes my proper phone number when I make outbound calls.
Thank you for that snippet Warren. I setup a different US DID and called that one via flowroute and the callerid worked. Previously I called a voip.ms toll-free number. I'll blame it on (lack of) carrier interoperability :) Good to know outbound callerid works without having to use magic SipAddHeader incantations.
Thanks! Patrick -- _____________________________________________________________________ -- 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
