pbx*CLI> <-- SIP read from Y.Y.Y.Y:6252: INVITE sip:31101@pbx.elephantoutlook.com SIP/2.0 Via: SIP/2.0/UDP 192.168.22.17:6252;branch=z9hG4bK-d87543-6e76826f93176173-1--d87543-;rport Max-Forwards: 70 Contact: To: From: "joe Webster <3149>";tag=42335900 Call-ID: f7573803537aca4f@b2JzZXJ2ZXI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="joelaptop",realm="asterisk",nonce="58f0ff85",uri="sip:31101@pbx.elephantoutlook.com",response="417fc459dd52345ac058e8da5a23f5c5",algorithm=MD5 Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 301 v=0 o=- 70358654 70358725 IN IP4 192.168.22.17 s=eyeBeam c=IN IP4 192.168.22.17 t=0 0 m=audio 6254 RTP/AVP 100 6 0 8 3 18 97 5 101 a=alt:1 1 : 9B24FA22 0000001C 192.168.22.17 6254 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (14 headers 12 lines)--- Using INVITE request as basis request - f7573803537aca4f@b2JzZXJ2ZXI. Sending to 192.168.22.17 : 6252 (NAT) Found user 'joelaptop' Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 5 Found RTP audio format 101 Peer audio RTP is at port 192.168.22.17:6254 Peer video RTP is at port 192.168.22.17:65535 Found description format speex Found description format speex Found description format telephone-event Capabilities: us - 0x80006 (gsm|ulaw|h263), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 31101 in local (domain pbx.elephantoutlook.com) list_route: hop: Transmitting (NAT) to Y.Y.Y.Y:6252: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.22.17:6252;branch=z9hG4bK-d87543-6e76826f93176173-1--d87543-;received=Y.Y.Y.Y;rport=6252 From: "joe Webster <3149>";tag=42335900 To: Call-ID: f7573803537aca4f@b2JzZXJ2ZXI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/joelaptop-b6a095f0", "Local/moh@local||tT") in new stack -- Called moh@local -- Executing WaitMusicOnHold("Local/moh@local-df08,2", "300") in new stack We're at 66.192.107.225 port 16142 Video is at 66.192.107.225 port 16434 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to Y.Y.Y.Y:6252: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.22.17:6252;branch=z9hG4bK-d87543-6e76826f93176173-1--d87543-;received=Y.Y.Y.Y;rport=6252 From: "joe Webster <3149>";tag=42335900 To: ;tag=as1098c46b Call-ID: f7573803537aca4f@b2JzZXJ2ZXI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 28675 28675 IN IP4 66.192.107.225 s=session c=IN IP4 66.192.107.225 t=0 0 m=audio 16142 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Started music on hold, class 'default', on channel 'Local/moh@local-df08,2' pbx*CLI>