Good afternoon I've read the document on bridging interfaces [1], reviewed the redirection part of the residential script & I think I understand the mhomed/manual methods etc but I'm really struggling with this problem.
My OpenSIPS server has a NAT'ed public IP and an internal IP on separate interfaces. Phones register via the public IP and I have an Asterisk server connected via the internal IP for voicemail etc: I'm using topology_hiding/match() for the initial call & SIP/RTP for a working call from one phone to another goes: PhoneA<-->NATedPublicIP<-->PhoneB If a call results in a 603 for example, the SIP is routed to Asterisk OK using the internal IP's but the media connection on the OpenSIPS end is still it's public IP so audio fails. How do I have OpenSIPS re-route the media so that in order to reach Asterisk it flows via the internal IP like this? My media server is rtpengine. Phone<-->NATedPublicIP<-->InternalIP<-->AsteriskInternal [1] https://blog.opensips.org/2018/09/04/sip-bridging-over-multiple-interfaces/ Many thanks! Mark
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