Hi All,
I'm new to MiVoB and to this forum, and very glad to be here.
I have a setup with MiVoB is connected to a carrier SIP Trunk without an MBG.
When I receive a call from pstn and then try an attended transfer back out to pstn, the call connects, but audio is lost.
My MiVoB and deskphone used to initiate the transfer are behind NAT.
I suspect I have problem either with nat/firewall or with my MiVoB setup/architecture.
Since MiVoB doesnt anchor the media in itself - I am wondering that which RTP endpoint should I consider is connecting the two call audio streams together with the attended transfer without MBG? I consider options: Lan phone, upstream pstn switch, other e.g. SBC.
I have recognized that the endpoint that connects the RTPs might affect the firewall configuration.
Thank you in advance.