SIP Debugging Enabled for IP: 10.0.1.40:5061 -- Executing [901489574204@handsets:1] Dial("SIP/3101-081e1220", "SIP/sipuraline1/901489574204|60") in new stack Audio is at 10.0.1.21 port 11784 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.1.40:5061: INVITE sip:901489574204@10.0.1.40:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK686e3713;rport From: "Handset 1" ;tag=as777e8f50 To: Contact: Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 21 Dec 2007 13:44:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 234 v=0 o=root 23787 23787 IN IP4 10.0.1.21 s=session c=IN IP4 10.0.1.21 t=0 0 m=audio 11784 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - =ptime:20> a=sendrecv --- -- Called sipuraline1/901489574204 akira*CLI> <--- SIP read from 10.0.1.40:5061 ---> SIP/2.0 100 Trying To: From: "Handset 1" ;tag=as777e8f50 Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK686e3713 Server: Linksys/SPA3102-3.3.6(GW) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- akira*CLI> <--- SIP read from 10.0.1.40:5061 ---> SIP/2.0 180 Ringing To: ;tag=20bea138f1fb2968i1 From: "Handset 1" ;tag=as777e8f50 Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK686e3713 Server: Linksys/SPA3102-3.3.6(GW) Remote-Party-ID: sipuraline1 ;screen=yes;party=called Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/sipuraline1-081e5188 is ringing akira*CLI> <--- SIP read from 10.0.1.40:5061 ---> SIP/2.0 200 OK To: ;tag=20bea138f1fb2968i1 From: "Handset 1" ;tag=as777e8f50 Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK686e3713 Contact: sipuraline1 Server: Linksys/SPA3102-3.3.6(GW) Remote-Party-ID: sipuraline1 ;screen=yes;party=called Content-Length: 247 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 190145 190145 IN IP4 10.0.1.40 s=- c=IN IP4 10.0.1.40 t=0 0 m=audio 16424 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 10.0.1.40:16424 Found description format PCMA for ID 8 Found description format NSE for ID 100 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.1.40:16424 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.1.40, port 5061 Transmitting (no NAT) to 10.0.1.40:5061: ACK sip:901489574204@10.0.1.40:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK7a0e5a42;rport From: "Handset 1" ;tag=as777e8f50 To: ;tag=20bea138f1fb2968i1 Contact: Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/sipuraline1-081e5188 answered SIP/3101-081e1220 -- Packet2Packet bridging SIP/3101-081e1220 and SIP/sipuraline1-081e5188 Scheduling destruction of SIP dialog '526dc357589aff4f601ca148086f39d9@10.0.1.21' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.1.40, port 5061 Reliably Transmitting (no NAT) to 10.0.1.40:5061: BYE sip:901489574204@10.0.1.40:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK39ba3204;rport From: "Handset 1" ;tag=as777e8f50 To: ;tag=20bea138f1fb2968i1 Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (handsets, 901489574204, 1) exited non-zero on 'SIP/3101-081e1220' akira*CLI> <--- SIP read from 10.0.1.40:5061 ---> SIP/2.0 200 OK To: ;tag=20bea138f1fb2968i1 From: "Handset 1" ;tag=as777e8f50 Call-ID: 526dc357589aff4f601ca148086f39d9@10.0.1.21 CSeq: 103 BYE Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK39ba3204 Server: Linksys/SPA3102-3.3.6(GW) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '526dc357589aff4f601ca148086f39d9@10.0.1.21' Method: INVITE Reliably Transmitting (no NAT) to 10.0.1.40:5061: OPTIONS sip:10.0.1.40:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK50290e8a;rport From: "asterisk" ;tag=as2c1f1d05 To: Contact: Call-ID: 7ee923d021d82d813cbd467c78aac9a7@10.0.1.21 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 21 Dec 2007 13:45:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- akira*CLI> <--- SIP read from 10.0.1.40:5061 ---> SIP/2.0 200 OK To: ;tag=b328fe0c274b6cc7i1 From: "asterisk" ;tag=as2c1f1d05 Call-ID: 7ee923d021d82d813cbd467c78aac9a7@10.0.1.21 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK50290e8a Server: Linksys/SPA3102-3.3.6(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '7ee923d021d82d813cbd467c78aac9a7@10.0.1.21' Method: OPTIONS akira*CLI>