Hey Steve, I tried to implement what you suggested and posted the console output to pastbin (14854). I forgot to add my dialplan xml, so I've pasted it here, since it's only a few lines. Sorry for pasting in the body of my email. Mike <extension name="sipexten"> <condition field="destination_number" expression="^(10[0-9][0-9])$"> <action application="export" data="sip_h_Subject" /> <action application="set" data="call_timeout=180"/> <action application="log" data="crit dialing extension $1" /> <action application="bridge" data="user/$1@${domain_name}" /> </condition> </extension> -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Wednesday, December 22, 2010 10:57 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Dropping the Subject: header I assume you're doing a bridge? FreeSWITCH is a b2bua... meaning the call going into FreeSWITCH is a separate call from the one going out (it just links the media on the 2 calls together to bridge them). That's why the variable's not being copied across by default. It is however possible to tell FreeSWITCH to copy a variable from the A-leg to the B-leg, and any non-standard SIP headers can be read and written using the sip_h_ variable prefix. Try this: <extension name="..."> <condition ...> ... <action application="export" data="sip_h_Subject" /> <action application="bridge" data="..." /> ... </condition> </extension> The subject should be in the sip_h_Subject variable on the A-leg, the above will export (copy) that variable onto the B-leg and it should then appear in the outgoing invite. -Steve On 22 December 2010 15:45, Kane, Michael (mkane02) <mkane02@harris.com> wrote:
Hello all, we are developing a custom app that will require sending a Subject: header (with a specific value) with the initial invite to trigger additional calling features and behavior. What we're finding when we send the Invite, is that FS is dropping the Subject: header and continues normal call processing. Is there a configuration parameter that allows the header to pass through FS?
I submitted a SIP trace in pastbin and the reference number is 14853.
Mike
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