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Another thing to note that as of Asterisk version 1.6 the refer method for transferring calls is not supported in sipX. Please refer to Asterisk documentation regarding SIP Refer method used by Asterisk. An untested work around since transfers worked in Asterisk version 1.4 is to use the option in sip.conf called pedantic, which will either make Asterisk pay more attention to SIP call-IDs or not.
Ref http://www.voip-info.org/wiki/index.php?page_id=2768
pedantic = yes|no : Enable slow, pedantic checking of Call-ID:s, multiline SIP headers and URI-encoded headers. Default no (in Asterisk 1.8 default yes)
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