<--- SIP read from UDP:174.136.97.12:5060 ---> INVITE sip:5646@sip.covert.org SIP/2.0 Via: SIP/2.0/UDP 174.136.97.12:5060;branch=z9hG4bK5ff468b0;rport Max-Forwards: 70 From: "William F. Acker" ;tag=as76ee3252 To: Contact: Call-ID: 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 CSeq: 102 INVITE User-Agent: The Octothorp Asterisk PBX Date: Tue, 21 Aug 2012 00:38:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "William F. Acker" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 336 v=0 o=root 1550527772 1550527772 IN IP4 174.136.97.12 s=Asterisk PBX 10.7.0 c=IN IP4 174.136.97.12 t=0 0 m=audio 19826 RTP/AVP 9 0 5 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - --- (15 headers 15 lines) --- Sending to 174.136.97.12:5060 (NAT) Using INVITE request as basis request - 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 Found peer 'wacker' for '3825*295' from 174.136.97.12:5060 Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 5 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format DVI4 for ID 5 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g722|h263|testlaw), peer - audio=(ulaw|alaw|adpcm|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 174.136.97.12:19826 Looking for 5646 in inbound-wacker (domain sip.covert.org) list_route: hop: <--- Transmitting (no NAT) to 174.136.97.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 174.136.97.12:5060;branch=z9hG4bK5ff468b0;received=174.136.97.12;rport=5060 From: "William F. Acker" ;tag=as76ee3252 To: Call-ID: 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 CSeq: 102 INVITE Server: John Covert Private Household SIP Service Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [5646@inbound-wacker:1] NoOp("SIP/wacker-00000008", "") in new stack -- Executing [5646@inbound-wacker:2] Dial("SIP/wacker-00000008", "SIP/x28,120,m(5xbring)") in new stack -- Called SIP/x28 -- Started music on hold, class '5xbring', on SIP/wacker-00000008 Audio is at 25842 Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 174.136.97.12:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 174.136.97.12:5060;branch=z9hG4bK5ff468b0;received=174.136.97.12;rport=5060 From: "William F. Acker" ;tag=as76ee3252 To: ;tag=as74c34303 Call-ID: 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 CSeq: 102 INVITE Server: John Covert Private Household SIP Service Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 312 v=0 o=root 1442059699 1442059699 IN IP4 108.20.57.196 s=Asterisk PBX 10.7.0 c=IN IP4 108.20.57.196 t=0 0 m=audio 25842 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/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 <------------> -- SIP/x28-00000009 is ringing -- SIP/x28-00000009 is ringing [Aug 20 20:38:34] NOTICE[24675]: chan_sip.c:13531 sip_reregister: -- Re-registration for 16789357030@sphone.vopr.vonage.net [Aug 20 20:38:34] NOTICE[24675]: chan_sip.c:21527 handle_response_register: Outbound Registration: Expiry for sphone.vopr.vonage.net is 20 sec (Scheduling reregistration in 16 s) -- SIP/x28-00000009 is ringing -- SIP/x28-00000009 is ringing -- SIP/x28-00000009 answered SIP/wacker-00000008 -- Stopped music on hold on SIP/wacker-00000008 Audio is at 25842 Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 174.136.97.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 174.136.97.12:5060;branch=z9hG4bK5ff468b0;received=174.136.97.12;rport=5060 From: "William F. Acker" ;tag=as76ee3252 To: ;tag=as74c34303 Call-ID: 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 CSeq: 102 INVITE Server: John Covert Private Household SIP Service Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Remote-Party-ID: "Snom 28" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 312 v=0 o=root 1442059699 1442059700 IN IP4 108.20.57.196 s=Asterisk PBX 10.7.0 c=IN IP4 108.20.57.196 t=0 0 m=audio 25842 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/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 -- Locally bridging SIP/wacker-00000008 and SIP/x28-00000009 <--- SIP read from UDP:174.136.97.12:5060 ---> ACK sip:5646@108.20.57.196:5060 SIP/2.0 Via: SIP/2.0/UDP 174.136.97.12:5060;branch=z9hG4bK32f68b7d;rport Max-Forwards: 70 From: "William F. Acker" ;tag=as76ee3252 To: ;tag=as74c34303 Contact: Call-ID: 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 CSeq: 102 ACK User-Agent: The Octothorp Asterisk PBX Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:174.136.97.12:5060 ---> INVITE sip:5646@108.20.57.196:5060 SIP/2.0 Via: SIP/2.0/UDP 174.136.97.12:5060;branch=z9hG4bK298afa71;rport Max-Forwards: 70 From: "William F. Acker" ;tag=as76ee3252 To: ;tag=as74c34303 Contact: Call-ID: 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 CSeq: 103 INVITE User-Agent: The Octothorp Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "William F. Acker" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 312 v=0 o=root 1550527772 1550527773 IN IP4 173.164.32.85 s=Asterisk PBX 10.7.0 c=IN IP4 173.164.32.85 t=0 0 m=audio 63548 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/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 <-------------> --- (14 headers 14 lines) --- Sending to 174.136.97.12:5060 (no NAT) Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g722|h263|testlaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 173.164.32.85:63548 <--- Transmitting (no NAT) to 174.136.97.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 174.136.97.12:5060;branch=z9hG4bK298afa71;received=174.136.97.12;rport=5060 From: "William F. Acker" ;tag=as76ee3252 To: ;tag=as74c34303 Call-ID: 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 CSeq: 103 INVITE Server: John Covert Private Household SIP Service Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 25842 Adding codec 100012 (g722) to SDP Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 174.136.97.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 174.136.97.12:5060;branch=z9hG4bK298afa71;received=174.136.97.12;rport=5060 From: "William F. Acker" ;tag=as76ee3252 To: ;tag=as74c34303 Call-ID: 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 CSeq: 103 INVITE Server: John Covert Private Household SIP Service Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 312 v=0 o=root 1442059699 1442059701 IN IP4 108.20.57.196 s=Asterisk PBX 10.7.0 c=IN IP4 108.20.57.196 t=0 0 m=audio 25842 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/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 <------------> <--- SIP read from UDP:174.136.97.12:5060 ---> ACK sip:5646@108.20.57.196:5060 SIP/2.0 Via: SIP/2.0/UDP 174.136.97.12:5060;branch=z9hG4bK6f951e94;rport Max-Forwards: 70 From: "William F. Acker" ;tag=as76ee3252 To: ;tag=as74c34303 Contact: Call-ID: 6d97e8135315fbd2556430f22ba425d9@174.136.97.12:5060 CSeq: 103 ACK User-Agent: The Octothorp Asterisk PBX Content-Length: 0 <-------------> # tcpdump -i en0 host 173.164.32.85 20:39:34.773201 IP asterisk.covert.org.25842 > snom320.octothorp.org.63548: UDP, length 172 20:39:34.777905 IP snom320.octothorp.org.63548 > asterisk.covert.org.25842: UDP, length 172 20:39:34.793204 IP asterisk.covert.org.25842 > snom320.octothorp.org.63548: UDP, length 172 20:39:34.798012 IP snom320.octothorp.org.63548 > asterisk.covert.org.25842: UDP, length 172 Got RTP packet from 192.168.0.41:65232 (type 09, seq 055336, ts 157279704, len 000160) Sent RTP P2P packet to 173.164.32.85:57528 (type 09, len 000160) Sent RTP packet to 192.168.0.41:65232 (type 09, seq 014072, ts 1739680376, len 000160) Got RTP packet from 192.168.0.41:65232 (type 09, seq 055337, ts 157279864, len 000160) Sent RTP P2P packet to 173.164.32.85:57528 (type 09, len 000160) Sent RTP packet to 192.168.0.41:65232 (type 09, seq 014073, ts 1739680536, len 000160) Got RTP packet from 192.168.0.41:65232 (type 09, seq 055338, ts 157280024, len 000160) Sent RTP P2P packet to 173.164.32.85:57528 (type 09, len 000160) Sent RTP packet to 192.168.0.41:65232 (type 09, seq 014074, ts 1739680696, len 000160) Got RTP packet from 192.168.0.41:65232 (type 09, seq 055339, ts 157280184, len 000160) Sent RTP P2P packet to 173.164.32.85:57528 (type 09, len 000160) Sent RTP packet to 192.168.0.41:65232 (type 09, seq 014075, ts 1739680856, len 000160) Got RTP packet from 192.168.0.41:65232 (type 09, seq 055340, ts 157280344, len 000160) Sent RTP P2P packet to 173.164.32.85:57528 (type 09, len 000160) Sent RTP packet to 192.168.0.41:65232 (type 09, seq 014076, ts 1739681016, len 000160)