Have recently installed some Snom phones into an office. Phones are natted and connect to a 1.4 server on a public IP
We can make outgoing calls, but are unable to answer incoming calls. The phone rings, but the call cannot be picked up. Other phones on other sites connected to the server are working perfectly. Looking at the SIP trace it appears the phone transmits: Sent to udp:193.33.xx.xx:5060 at 6/1/2011 11:49:20:868 (849 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 193.33.xx.xx:5060;branch=z9hG4bK6e82052c;rport=5060 From: "xx" <sip:[email protected]>;tag=as1b6fc27c To: <sip:[email protected]:25380>;tag=37gg1zu3wp Call-ID: [email protected] CSeq: 102 INVITE Contact: <sip:[email protected]:2048>;reg-id=1 User-Agent: snom300/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 220 v=0 o=root 641540583 641540584 IN IP4 192.168.4.19 s=call c=IN IP4 192.168.4.19 t=0 0 m=audio 52386 RTP/AVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv but it is never received by the server. Interestingly RINGING and REGISTER messages are working OK. The NAT router is out of our control. Are we looking at a SIP ALG getting in the way? Thanks, John -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
