Hi,
I have been working with Cisco VoIP since 2001 and Asterisk and freepbx ,trixbox, elastix since 2007.
I'm very new to the site so trying to bid as low as possible and get better reviews as much as possible.
That being said I would like to shed some light on the issues you might have:
1) may be the codecs you are having are not compatible with the other end ( first and very obvious thing in the error message)
2) the context might be wrong
3) you may get the answer by yourself just by debugging the sip messages ( that's what I'm going to do!)
if you think you still want to hire someone to do the job, kindly do consider me :)
just let me know when can have some chat to better understand the project,
regards,