>>>>> "MJ" == Matthew Jordan <mjor...@digium.com> writes:
MJ> That is incorrect. The sip_sendhtml callback will update the url MJ> stringfield on the SIP pvt. It then transmits a re-INVITE via MJ> transmit_reinvite_with_sdp. There was no re-INVITE, just the initial INVITE. And it did not have an Access-URL header. If Dial()'s url is only sent after the calle answers, it is of no value. The callee needs the information to decide whether to answer. MJ> "Test call" doesn't tell us much. Is the Access-URL header added to MJ> an outgoing re-INVITE? There was no re-INVITE. My test was a pstn call to my provider, passed to my asterisk-11's followme(). The Dial() made in the context followme() used looks like (with some names changed): exten => soft,1,Verbose(0,Dialing ${EXTEN} to ekiga) same => n,Dial(SIP/softek/${EXTEN},60,,https://jhcloos.com/sip) same => n,Hangup() MJ> As Dennis pointed out, you can add whatever header you want to your MJ> outgoing INVITE requests using SIPAddHeader or - in the PJSIP stack - MJ> PJSIP_HEADER. That includes the Access-URL header, with whatever MJ> contents you want. OK. Missed that last night. Thanks. MJ> The ability to add whatever header you want to your outbound INVITE MJ> requests is a much more powerful abstraction Agreed. Heartily. And it works nicely. I should have noticed that ☹. Thanks! -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev