asterisk users - Jul 2005 - INVITE/REFER with only 2 ends on asterisk

hello,

i'm currently using asterisk (cvs-head) as PSTN gateway. the routing
logic is mostly done in OpenSER. the problem is that i'm not able to
transfer calls between the PSTN and another SIP peer (when the PSTN<>SIP
connection goes over asterisk but the SIP<>SIP connection does not).

there are 2 possibilities for asterisk to be part of the transfer:

1) asterisk receives the REFER msg. in this case asterisk logs the
following error msg:
"Supervised transfer requested, but unable to find callid '%s'.
Both
legs must reside on Asterisk box to transfer at this time."

2) the REFER goes to the other peer - asterisk just receives the INVITE:
this basically works but it seems that asterisk ignores the 'replaces'
header. therefore a new call is initiated instead of a transfer of the
current call.

i've checked the STABLE src and it seems to be affected by this problem
too (i haven't tested it - just skimmed through the src).

the only solution for transfers to work seems to be to route all calls
over asterisk - which is a solution i'd like to avoid because of
scalability and compability (it breaks ICE) issues.

any other suggestion how to get transfers to work without routing _all_
the calls through asterisk?

/gst

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