<-------------> --- (0 headers 0 lines) Nat keepalive --- internal*CLI> <--- SIP read from 203.166.6.160:5060 ---> INVITE sip:61280642141@121.44.8.82:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 203.166.6.160;branch=z9hG4bK728d.2cc77ca8c8b89410e7c306462dfa63a9.0 Via: SIP/2.0/UDP 203.166.6.160:5061;branch=z9hG4bK6b610698c958de075e4490d1c87890bc;rport=5061 Max-Forwards: 16 From: ;tag=7062e6f9882f71ba07ba6015b644d39d To: Call-ID: 63F046B5-11711DD-8221F052-82EB7ED6@202.85.241.123 CSeq: 200 INVITE Contact: Anonymous Expires: 300 User-Agent: Sippy cisco-GUID: 1676451116-18289117-2183065682-2196471510 h323-conf-id: 1676451116-18289117-2183065682-2196471510 Content-Length: 385 Content-Type: application/sdp v=0 o=Sippy 145298092 0 IN IP4 203.166.6.160 s=SIP Call t=0 0 m=audio 16984 RTP/AVP 18 3 4 8 0 2 101 19 c=IN IP4 202.85.241.123 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 <-------------> --- (16 headers 17 lines) --- Sending to 203.166.6.160 : 5060 (no NAT) Using INVITE request as basis request - 63F046B5-11711DD-8221F052-82EB7ED6@202.85.241.123 Found peer 'Pennytel-out' Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 101 Found RTP audio format 19 Peer audio RTP is at port 202.85.241.123:16984 Found audio description format G729 for ID 18 Found audio description format GSM for ID 3 Found audio description format G723 for ID 4 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format telephone-event for ID 101 Found audio description format CN for ID 19 Capabilities: us - 0x0 (nothing), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) <--- Reliably Transmitting (NAT) to 203.166.6.160:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 203.166.6.160;branch=z9hG4bK728d.2cc77ca8c8b89410e7c306462dfa63a9.0;received=203.166.6.160 Via: SIP/2.0/UDP 203.166.6.160:5061;branch=z9hG4bK6b610698c958de075e4490d1c87890bc;rport=5061 From: ;tag=7062e6f9882f71ba07ba6015b644d39d To: ;tag=as68927a69 Call-ID: 63F046B5-11711DD-8221F052-82EB7ED6@202.85.241.123 CSeq: 200 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0