[Asterisk-Users] Unable to prevent SIP to SIP calls from removing Asterisk from Media path

[Johann]

Due to problems with SIP transfers and agents, we are using blind 
transfers in asterisk(# key) for all calls.  With 1.2.1, Asterisk is 
doing a native bridge regardless.

Dial(SIP/phone,,to)

Using the above dial string and I see on the console that Asterisk is 
attempting a native bridge.  This breaks the blind transfers :(

Also tried putting, the below in sip.conf for the phones without success:
canreinvite=no

Any advice?

--johann