[asterisk-dev] dial url with sip

[James Cloos]

Looking at app_dial.c and chan_sip.c, I get the impression that the url
in a dial string cannot get sent as part of the sip INVITE, yes?

(I base that on sip_sendhtml().)

Am I reading chan_sip correctly?  Will I need to change sip_sendhtml()
to send the url as part of the INVITE?

A test call shows no url is sent.

(I also see that in 12 and trunk chan_pjsip does not have a send_html
entry in its chan_pjsip_tech structure, and is therefore less capable.)

My understanding is that some sip phones will fetch and display a url
when INVITEd, and I'd like to use that to show the callee more data
about the incoming call, such as the remote sip proxy/endpoint, the
details about the INVITEd number, et cetera.  In particular, I want to
do this will dials generated as a result of followme, queuesand the
like.

That will only work if the url is part of the INVITE from ast to the phone.

-JimC
--
James Cloos <cloos at jhcloos.com>         OpenPGP: 0x997A9F17ED7DAEA6