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
