[Freeswitch-users] Dropping the Subject: header

[Kane, Michael (mkane02)]

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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at 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 at 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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