Just purchased a 6410. Here is my scenario. We receive SIP trunks from our ISP. We have two dialers that make calls and then hand the call to our phones. We used to have the dialers go through our Cisco ASA. When attempting to replace that Cisco ASA with a Cisco Meraki our dialers could no longer make calls. After inspecting a PCAP this was because the Meraki does not replace the internal IP address(192.168.40.x) with the Public facing address. The SIP trunk provider requires the source address in the packet header to come from the public address. So what would happen is the SIP connection would take place but we would get 1 way audio because the RTP stream was not using the public address.
From what I understand this has to do with SIP ALG. The 6410 does not appear to have the capability to enable nor disable SIP ALG. Is that expected?
Can I expect the Adtran to replace the Source Address during NAT with the public address in the RTP stream?