We didn't currently support that but I added a patch to trunk that should let you do it. sip_outgoing_call_id variable set on a new channel should let you choose your own call id. on the a leg ${sip_call_id} could be passed to the b leg one of 2 ways. using the "export" app on the A leg: one call before the bridge <action application="export" data="sip_outgoing_call_id=${sip_call_id}"/> <action application="bridge" data="sofia/default/foo@bar.com"/> using the originate syntax in the bridge. <action application="bridge" data="{sip_outgoing_call_id=${sip_call_id}}sofia/default/foo@bar.com"/> On Sat, Jun 7, 2008 at 12:31 AM, Jed Stafford <jedsta@gmail.com> wrote:

I'm working to get FreeSwitch working behind a SIP load balancer. Using the sip-ip param in a profile I'm able to make the outbound calls appear to come from the load balancer IP address. Debugging the sip messages, this all appears to be working correctly.

However the load balancer determines which machine to send the calls too based on the SIP Call-ID. We want FreeSwitch in the middle of the call for billing and other purposes. But i'd like FreeSwitch to use the same SIP Call-ID that was sent to it when it initiates the next leg of the call. The hash on the load balancer only cares about everything proceeding the @ sign in the sip call-id. Is there a function in the dialplan to copy the source call-id and use it for the destination leg?

Hopefully I've explained this well enough, any ideas would be helpful.

Regards,

-Jed

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