[asterisk-dev] dial url with sip

[James Cloos]

>>>>> "MJ" == Matthew Jordan <mjordan at 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 <cloos at jhcloos.com>         OpenPGP: 0x997A9F17ED7DAEA6