rune Posted September 1, 2007 Report Share Posted September 1, 2007 Hi, I have a problem with one way audio when I recieve incomming call on the trunk. My setup: The pbxnsip is placed on a LAN behind a DLINK 603 router. On the same network I have 3 sip-phones. The pbxnsip has a registration on musimi.dk. The trunk is configured to forward incomming call to extention "2222". The 3 sip phones has each a registration on the pbxnsip (phone number 1111,2222,3333). If I make an incomming call to the musimi account (the trunk on pbxnsip), phone 2222 alerts, when I answer there is no TX audio from phone 2222 to the external device. Any hints why? Does the pbxnsip requires a public IP address to route the RTP correctly? What NAT traversal features does pbxnsip have? I cannot find a setting for rport or ice. I have made an wireshark trace and audio from the extention unit is routed correctly to the pbxnsip (I have checked ports, ip-addresses). If I move the trunk SIP account to another phone on the LAN and make an incomming call there is audio both ways (I have enabled rport on the device). Likewise if phone 1111 calls 2222 (internal call on the pbx) audio is also working fine. Another issue is that if the trunk registration has multiple bindings on the external SIP server (in this case musimi) the pbx cancelles the incomming call when the all is answered. Quote Link to comment Share on other sites More sharing options...
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