[asterisk-dev] dial url with sip

[Dennis Guse]

If you are in control of the SIP-Phone, you could pass additional
information via SIPAddHeader in your dialplan.
On Jun 2, 2014 10:33 AM, "James Cloos" <cloos at jhcloos.com> wrote:

> 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
>
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