[Asterisk-Users] Error in Incoming SIP call

[Olle E. Johansson]

--unique-boundary-1
Content-Type: application/ISUP;version=cp10isup;base=etsi121
Content-Disposition: signal;handling=optional

01 07 02 70 00 02 01 03 09 02 0a 00 0a 07 03 13 15 44 12 01 20 04 08 83 10 15 74
77 11 11 0f 06 01 10 00
--unique-boundary-1

Hi!

Content-type: application/ISUP

---> http://www.ietf.org/rfc/rfc3204.txt
    This document describes MIME types for application/ISUP and
    application/QSIG objects for use in SIP applications, according to
    the rules defined in RFC 2048.  These types can be used to identify
    ISUP and QSIG objects within a SIP message such as INVITE or INFO, as
    might be implemented when using SIP in an environment where part of
    the call involves interworking to the PSTN.
-------

Interesting, but understandable if it's a softswitch. Haven't seen that one before.
Seems like we need a small change in the MIME handling of the chan_sip to handle
MIME multipart message bodies. Right now, the message body is parses from top down,
regardless of what's in it (as far as I can understand the source).

Thank your for the debug output! There's always something new to learn, isn't it?

/O