[asterisk-dev] dial url with sip

[Matthew Jordan]

On Mon, Jun 2, 2014 at 4:14 PM, James Cloos <cloos at jhcloos.com> wrote:
>>>>>> "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.

That is what it does:

 * \brief dial() & retrydial() - Trivial application to dial a channel
and send an URL on answer

So yes, the URL option probably isn't of much use to you.

Whether or not it sent the re-INVITE: we'd have to investigate a lot
further, which sounds like it isn't worth it.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org