[Asterisk-Users] SIP/ShoreTel REFER support

[Magnus Kelly]

Hello All,
Here's the problem, we have happily set up several Asterisk servers to offer
commercial service in the UK, our wholesale SIP termination partner
(Magrathea - use SER/CiscoGW to provide us the service on a public IP
address) - till now we have used Asterisk to connect clients on private IP's
with Asterisk doing the required conversion for SIP/IAX between public and
private IP's.

The current issue is that we have recently agreed to support ShoreTel PBX's
with their new SIP trunk feature, and in staging the first install we have
found that certain features (blind transfer) require support for both SIP
Refer and Refer Replace - which are not supported by the current VoIP
provider SER config. (For some good reasons as they use public IP's)

So the challenge is to quickly work out the possibility of either adding a
SER setup in-between the ShoreTel PBX and the VoIP provider SER unit or
preferably finding a way to use one of our current asterisk servers to
provide support for this need.
The intent of this setup is to both allow for NAT - E.g. use private IP's
for the ShoreTel system and public Ip for the VoIP provider, as well as
ensuring that the local Asterisk/SER server supports the required Refer and
Refer replace commands to allow the ShoreTel PBX to be able to offer blind
transfer support.
ShoreTel uses the below call control steps during a transfer with the
current architecture:

.   Blind transfer: A calls B. A puts B on hold. A sends a REFER to B
transferring it to C.

.   Consult transfer: A calls B. A puts B on hold. A calls C. A puts C on
hold. A sends REFER to C transferring it to B.

ShoreTel architecture uses SIP REFER method for blind transfers and SIP
REFER with Replaces header to do consult transfers.

This means that since (For NAT reasons) our SIP.conf has two contexts - Sip
trunk and ShoreTel trunk both have reinvite=no (also to maintain billing
records) the SIP Refer functions are not working as planned or hoped.Or
Refer is not supported?

My problems are:
a) My friend Google has little to offer in exactly which RFC's Asterisk
supports (particularly as recently Google does not search correctly the list
archives?) - Is the SIP Refer function supported?
b) Very short timetable to deliver the working solution - 1
week-Particularly if we have to plunge into adding SER to the mix - Steep
learning curve with SER? - as some (most?) of IpTels web site is down?

Can any one offer guidance on whether my proposed solution will work and
share any tips on problems I should be aware of?

If any one is interested in taking this on as an Easter project for minor
commercial reward - email me off list (magnus at mcomwifi dot net)

If this is the wrong list for this type of thing - Apologies

Thanks
Magnus