We didn't currently support that but I added a patch to trunk that should let you do it.<br><br>sip_outgoing_call_id variable set on a new channel should let you choose your own call id.<br><br>on the a leg ${sip_call_id} could be passed to the b leg one of 2 ways.<br>
<br>using the "export" app on the A leg: one call before the bridge<br><br><action application="export" data="sip_outgoing_call_id=${sip_call_id}"/><br><action application="bridge" data="sofia/default/<a href="mailto:foo@bar.com">foo@bar.com</a>"/><br>
<br>using the originate syntax in the bridge.<br><br><action application="bridge" data="{sip_outgoing_call_id=${sip_call_id}}sofia/default/<a href="mailto:foo@bar.com">foo@bar.com</a>"/><br>
<br><br><br><br><div class="gmail_quote">On Sat, Jun 7, 2008 at 12:31 AM, Jed Stafford <<a href="mailto:jedsta@gmail.com">jedsta@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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.<br>
<br>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?<br>
<br>Hopefully I've explained this well enough, any ideas would be helpful.<br><br>Regards,<br><font color="#888888"><br>-Jed<br><br>
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