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" <cl...@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 <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 >
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