I have an untested solution to this issue. Note: I am using JavaScript code examples: The following will do the "divert"/"transfer" if you have a session. session.execute("deflect", dfltstr); When Sip is involved, FreeSwitch when using a SIP "Call Transfer" which asks some upstream system to send the call to some other system. When doing this, a new destination address can be supplied. This can include a "postd" parameter. I think postd stands for "post dial" and is a standard parameter that can be added to the destination address. RFC 3261, section 19.1.6 has the following example: Thus, tel:+358-555-1234567;postd=pp22 becomes sip:+358-555-1234567;postd=pp22@foo.com;user=phone There should be a way to add a postd string to the new destination address. So you can put the "user id" into the postd string. When the divert/transfer occurs, the new system will get the call as if it was an originated call. Looking at the destination string, the presence or absence of postd can determine how the call should be handled. If a postd is provided, the value can be used as the "user id". Note: I was looking at conferencing so planned to pass the conference information this way. Another note: I don't think that "user=phone" is necessary. The systems seem to be designed to work with it provided or not provided. If/When I find time, I will code this up as a JavaScript example and add it to the Wiki. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Two-FreeSWITCH-Box-IVR-Call-Tr... Sent from the freeswitch-users mailing list archive at Nabble.com.