Hello,

I have an issue with Mitel 3300 and Juniper SSG20
Plssssssssssss help.


There are two sites (Site A and B) that are connected by SSG 20 on each side via site-site VPN. We have SIP links coming into each end from the SIP provider which is NAT'ed to the Mitel controller. I can call extensions inernally across the sites and vice versa. However I'm having some issues:

1. Call comes in on site A, gets call attendant, request extension on the site B.
If someone answers there is no audio either direction.
If no one answers, caller just has dead air until the call dies.

2. A call comes in to the site A and is answered locally, then needs to be transferred to site B.
If done via conference the call is connected and communications works until the site A user hangs up, then both other users get dropped.
If done via transfer, the incoming caller gets dead air and the remote caller does get the ring but no audio

I have tried enabling SIp, etc. Pls help me to fix this issue.

Thanks
Sharu