*CLI> == Using SIP RTP CoS mark 5 -- Executing [2002@myusers:1] Dial("SIP/saghul1-00000006", "SIP/saghul2") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x1000 (g722) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.99.48:56164: INVITE sip:okpedxsv@192.168.99.48:56164 SIP/2.0 Via: SIP/2.0/UDP 192.168.99.46:5060;branch=z9hG4bK40d369ab Max-Forwards: 70 From: "saghul1" ;tag=as1ec6c567 To: Contact: Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r310240 Date: Thu, 10 Mar 2011 21:15:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 304 v=0 o=root 1744741950 1744741950 IN IP4 192.168.99.46 s=Asterisk PBX SVN-branch-1.8-r310240 c=IN IP4 192.168.99.46 t=0 0 m=audio 10796 RTP/AVP 9 8 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called saghul2 <--- SIP read from UDP:192.168.99.48:56164 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.99.46:5060;received=192.168.99.46;branch=z9hG4bK40d369ab Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 From: "saghul1" ;tag=as1ec6c567 To: CSeq: 102 INVITE Server: Blink Pro 1.0.5 (MacOSX) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.99.48:56164 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.99.46:5060;received=192.168.99.46;branch=z9hG4bK40d369ab Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 From: "saghul1" ;tag=as1ec6c567 To: ;tag=Cni5QwdMV5uNjE80R8ca68JA3TMmJe66 CSeq: 102 INVITE Server: Blink Pro 1.0.5 (MacOSX) Contact: Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/saghul2-00000007 is ringing <--- SIP read from UDP:192.168.99.48:56164 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.99.46:5060;received=192.168.99.46;branch=z9hG4bK40d369ab Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 From: "saghul1" ;tag=as1ec6c567 To: ;tag=Cni5QwdMV5uNjE80R8ca68JA3TMmJe66 CSeq: 102 INVITE Server: Blink Pro 1.0.5 (MacOSX) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: Supported: 100rel, norefersub Content-Type: application/sdp Content-Length: 241 v=0 o=- 3508780434 3508780435 IN IP4 192.168.99.48 s=Blink Pro 1.0.5 (MacOSX) c=IN IP4 192.168.99.48 t=0 0 m=audio 50006 RTP/AVP 9 101 a=rtcp:50007 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (12 headers 11 lines) --- Found RTP audio format 9 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - 0x1008 (alaw|g722), peer - audio=0x1000 (g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1000 (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.99.48:50006 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.99.48:56164 Transmitting (no NAT) to 192.168.99.48:56164: ACK sip:okpedxsv@192.168.99.48:56164 SIP/2.0 Via: SIP/2.0/UDP 192.168.99.46:5060;branch=z9hG4bK327d05cb Max-Forwards: 70 From: "saghul1" ;tag=as1ec6c567 To: ;tag=Cni5QwdMV5uNjE80R8ca68JA3TMmJe66 Contact: Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r310240 Content-Length: 0 --- -- SIP/saghul2-00000007 answered SIP/saghul1-00000006 -- Remotely bridging SIP/saghul1-00000006 and SIP/saghul2-00000007 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.99.48:56164 Audio is at 5060 Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.99.48:56164: INVITE sip:okpedxsv@192.168.99.48:56164 SIP/2.0 Via: SIP/2.0/UDP 192.168.99.46:5060;branch=z9hG4bK6f374ec6 Max-Forwards: 70 From: "saghul1" ;tag=as1ec6c567 To: ;tag=Cni5QwdMV5uNjE80R8ca68JA3TMmJe66 Contact: Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r310240 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 280 v=0 o=root 1744741950 1744741951 IN IP4 192.168.99.50 s=Asterisk PBX SVN-branch-1.8-r310240 c=IN IP4 192.168.99.50 t=0 0 m=audio 50296 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.99.48:56164 ---> SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.99.46:5060;received=192.168.99.46;branch=z9hG4bK6f374ec6 Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 From: "saghul1" ;tag=as1ec6c567 To: ;tag=Cni5QwdMV5uNjE80R8ca68JA3TMmJe66 CSeq: 103 INVITE Warning: 399 Blink Pro 1.0.5 (MacOSX) "Difference in contents of o= line" Server: Blink Pro 1.0.5 (MacOSX) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.99.48:56164 Transmitting (no NAT) to 192.168.99.48:56164: ACK sip:okpedxsv@192.168.99.48:56164 SIP/2.0 Via: SIP/2.0/UDP 192.168.99.46:5060;branch=z9hG4bK6f374ec6 Max-Forwards: 70 From: "saghul1" ;tag=as1ec6c567 To: ;tag=Cni5QwdMV5uNjE80R8ca68JA3TMmJe66 Contact: Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r310240 Content-Length: 0 --- Scheduling destruction of SIP dialog '6ad9402e6c8278520613834a59336134@192.168.99.46:5060' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.99.48:56164 Reliably Transmitting (no NAT) to 192.168.99.48:56164: BYE sip:okpedxsv@192.168.99.48:56164 SIP/2.0 Via: SIP/2.0/UDP 192.168.99.46:5060;branch=z9hG4bK41811cd9 Max-Forwards: 70 From: "saghul1" ;tag=as1ec6c567 To: ;tag=Cni5QwdMV5uNjE80R8ca68JA3TMmJe66 Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r310240 X-Asterisk-HangupCause: Bearer capability not available X-Asterisk-HangupCauseCode: 58 Content-Length: 0 --- == Spawn extension (myusers, 2002, 1) exited non-zero on 'SIP/saghul1-00000006' <--- SIP read from UDP:192.168.99.48:56164 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.99.46:5060;received=192.168.99.46;branch=z9hG4bK41811cd9 Call-ID: 6ad9402e6c8278520613834a59336134@192.168.99.46:5060 From: "saghul1" ;tag=as1ec6c567 To: ;tag=Cni5QwdMV5uNjE80R8ca68JA3TMmJe66 CSeq: 104 BYE Server: Blink Pro 1.0.5 (MacOSX) Content-Length: 0 <-------------> --- (8 headers 0 lines) ---