SIP Debugging Enabled for IP: 192.168.115.135:5060 telefon*CLI> <--- SIP read from 192.168.115.135:5060 ---> SIP/2.0 200 OK To: ;tag=3ade143c2a5d0947i0 From: "XXX 0000000" ;tag=as6255c919 Call-ID: 11409ad50c580ef4333a9d1f2ede6889@192.168.115.120 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.115.120:5060;branch=z9hG4bK66885a6d Contact: "xxxxxx" Server: Linksys/SPA922-4.1.18 Content-Length: 212 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 474580 474580 IN IP4 192.168.115.135 s=- c=IN IP4 192.168.115.135 t=0 0 m=audio 16418 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.115.135:16418 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.115.135:16418 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.115.135, port 5060 Transmitting (no NAT) to 192.168.115.135:5060: ACK sip:321@192.168.115.135:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.115.120:5060;branch=z9hG4bK7c624824;rport From: "XXX 0000000" ;tag=as6255c919 To: ;tag=3ade143c2a5d0947i0 Contact: Call-ID: 11409ad50c580ef4333a9d1f2ede6889@192.168.115.120 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0