[asterisk-dev] dial url with sip

2014-06-02 Thread 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

Re: [asterisk-dev] dial url with sip

2014-06-02 Thread 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 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

Re: [asterisk-dev] dial url with sip

2014-06-02 Thread James Cloos
MJ == Matthew Jordan mjor...@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

Re: [asterisk-dev] dial url with sip

2014-06-02 Thread James Cloos
DG == Dennis Guse dennis.g...@qu.tu-berlin.de writes: DG If you are in control of the SIP-Phone, you could pass additional DG information via SIPAddHeader in your dialplan. Thanks. That turns out to be exactly what I wanted. I've spent the last few years at a lower level, including when

Re: [asterisk-dev] dial url with sip

2014-06-02 Thread Matthew Jordan
On Mon, Jun 2, 2014 at 4:14 PM, James Cloos cl...@jhcloos.com wrote: MJ == Matthew Jordan mjor...@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