akira*CLI> <--- SIP read from 10.0.1.41:5060 ---> INVITE sip:901489574204@10.0.1.21 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-194e1a59 From: ;tag=b2137b34817f28b9o0 To: Call-ID: 30c12d10-963bc08d@10.0.1.41 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="3352271d",uri="sip:901489574204@10.0.1.21",algorithm=MD5,response="39814b89d35d1938a3900fb67b17c257" Contact: Expires: 240 User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 391 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 1333214 1333214 IN IP4 10.0.1.41 s=- c=IN IP4 10.0.1.41 t=0 0 m=audio 16394 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (14 headers 18 lines) --- Sending to 10.0.1.41 : 5060 (no NAT) Using INVITE request as basis request - 30c12d10-963bc08d@10.0.1.41 Found user '3101' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 10.0.1.41:16394 Found description format PCMU for ID 0 Found description format G726-32 for ID 2 Found description format G723 for ID 4 Found description format PCMA for ID 8 Found description format G729a for ID 18 Found description format G726-40 for ID 96 Found description format G726-24 for ID 97 Found description format G726-16 for ID 98 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/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 10.0.1.41:16394 Looking for 901489574204 in handsets (domain 10.0.1.21) list_route: hop: <--- Transmitting (no NAT) to 10.0.1.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-194e1a59;received=10.0.1.41 From: ;tag=b2137b34817f28b9o0 To: Call-ID: 30c12d10-963bc08d@10.0.1.41 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [901489574204@handsets:1] Dial("SIP/3101-081e1220", "SIP/sipuraline1/901489574204|60") in new stack Audio is at 10.0.1.21 port 18352 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=z9hG4bK671dcf5c;rport From: "Handset 1" ;tag=as7f28049e To: Contact: Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 21 Dec 2007 13:37:26 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 18352 RTP/AVP 8 101 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 sipuraline1/901489574204 akira*CLI> <--- SIP read from 10.0.1.40:5061 ---> SIP/2.0 100 Trying To: From: "Handset 1" ;tag=as7f28049e Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK671dcf5c 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=52de3dd8aa3e8526i1 From: "Handset 1" ;tag=as7f28049e Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK671dcf5c 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 <--- Transmitting (no NAT) to 10.0.1.41:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-194e1a59;received=10.0.1.41 From: ;tag=b2137b34817f28b9o0 To: ;tag=as2fd54815 Call-ID: 30c12d10-963bc08d@10.0.1.41 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> akira*CLI> <--- SIP read from 10.0.1.40:5061 ---> SIP/2.0 200 OK To: ;tag=52de3dd8aa3e8526i1 From: "Handset 1" ;tag=as7f28049e Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK671dcf5c 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=- 145055 145055 IN IP4 10.0.1.40 s=- c=IN IP4 10.0.1.40 t=0 0 m=audio 16422 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 10.0.1.40:16422 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 10.0.1.40:16422 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=z9hG4bK411b2875;rport From: "Handset 1" ;tag=as7f28049e To: ;tag=52de3dd8aa3e8526i1 Contact: Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/sipuraline1-081e5188 answered SIP/3101-081e1220 Audio is at 10.0.1.21 port 14836 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.0.1.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-194e1a59;received=10.0.1.41 From: ;tag=b2137b34817f28b9o0 To: ;tag=as2fd54815 Call-ID: 30c12d10-963bc08d@10.0.1.41 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: 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 14836 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/3101-081e1220 and SIP/sipuraline1-081e5188 akira*CLI> <--- SIP read from 10.0.1.41:5060 ---> ACK sip:901489574204@10.0.1.21 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-4eba0be2 From: ;tag=b2137b34817f28b9o0 To: ;tag=as2fd54815 Call-ID: 30c12d10-963bc08d@10.0.1.41 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="3352271d",uri="sip:901489574204@10.0.1.21",algorithm=MD5,response="4c430167f5abebad08a371728a5b4c81" Contact: User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 <-------------> --- (11 headers 0 lines) --- 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=z9hG4bK744752d0;rport From: "asterisk" ;tag=as15eb1f97 To: Contact: Call-ID: 6fe3b1f029f2e6766c4063176ea70da1@10.0.1.21 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 21 Dec 2007 13:37:29 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/2.0 486 Busy Here To: ;tag=b328fe0c274b6cc7i1 From: "asterisk" ;tag=as15eb1f97 Call-ID: 6fe3b1f029f2e6766c4063176ea70da1@10.0.1.21 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK744752d0 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 '6fe3b1f029f2e6766c4063176ea70da1@10.0.1.21' Method: OPTIONS akira*CLI> <--- SIP read from 10.0.1.41:5060 ---> BYE sip:901489574204@10.0.1.21 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-212b23c0 From: ;tag=b2137b34817f28b9o0 To: ;tag=as2fd54815 Call-ID: 30c12d10-963bc08d@10.0.1.41 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="3101",realm="asterisk",nonce="3352271d",uri="sip:901489574204@10.0.1.21",algorithm=MD5,response="8894196a8344d6f39f639cf049b9499a" User-Agent: Sipura/SPA921-4.1.10(b) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.0.1.41 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.0.1.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.1.41:5060;branch=z9hG4bK-212b23c0;received=10.0.1.41 From: ;tag=b2137b34817f28b9o0 To: ;tag=as2fd54815 Call-ID: 30c12d10-963bc08d@10.0.1.41 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5d4d04c40fe4fb5878cee7a43f3556bb@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=z9hG4bK175be556;rport From: "Handset 1" ;tag=as7f28049e To: ;tag=52de3dd8aa3e8526i1 Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@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=52de3dd8aa3e8526i1 From: "Handset 1" ;tag=as7f28049e Call-ID: 5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21 CSeq: 103 BYE Via: SIP/2.0/UDP 10.0.1.21:5060;branch=z9hG4bK175be556 Server: Linksys/SPA3102-3.3.6(GW) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '5d4d04c40fe4fb5878cee7a43f3556bb@10.0.1.21' Method: INVITE Really destroying SIP dialog '30c12d10-963bc08d@10.0.1.41' Method: BYE