{{Box Important|This page was written in 2005 and is a bit out of date. Most of the reasons why you would want to interconnect sipX with Asterisk have gone away as sipX has not only caught up in terms of supported features, but in more and more areas has taken the lead, is more robust and easier to use. We expect therefore that this page will be deleted early 2007 after we finish sipX release 3.8.|} -Still here as of Oct. 2014.
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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", "to", and "from" or just "call-IDs" alone.
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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