asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 49 (Unspecified) D 5060 UNKNOWN 48 (Unspecified) D 5060 UNKNOWN 51 (Unspecified) D 5060 UNKNOWN 45 (Unspecified) D 5060 UNKNOWN 44/44 192.168.0.44 D 5060 OK (16 ms) 43/43 192.168.0.43 D 2063 OK (34 ms) 42/42 192.168.0.42 D 2063 OK (99 ms) 7 sip peers [Monitored: 3 online, 4 offline Unmonitored: 0 online, 0 offline] asterisk*CLI> sip set debug on asterisk*CLI> <--- SIP read from TLS://192.168.0.43:2097 ---> INVITE sip:42@192.168.0.1;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.43:2063;branch=z9hG4bK-qceld7ydcw52;rport From: "43" ;tag=pj325n8gie To: Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 475 v=0 o=root 1565129441 1565129441 IN IP4 192.168.0.43 s=call c=IN IP4 192.168.0.43 t=0 0 m=audio 57626 RTP/AVP 8 0 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Sg344vMy93smKq7f22kFiKjwavjW5TeBVUDAgJSG a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv �@�޷� l�޷ <-------------> --- (18 headers 20 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.0.43 : 2097 (NAT) Using INVITE request as basis request - 3c27c0545608-82f3mst3zcpd Found user '43' for '43' <--- Reliably Transmitting (no NAT) to 192.168.0.43:2063 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.0.43:2063;branch=z9hG4bK-qceld7ydcw52;received=192.168.0.43;rport=2097 From: "43" ;tag=pj325n8gie To: ;tag=as77ba5dcc Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 1 INVITE User-Agent: FHVOIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk-server", nonce="4aa19ad4" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c27c0545608-82f3mst3zcpd' in 32000 ms (Method: INVITE) asterisk*CLI> <--- SIP read from TLS://192.168.0.43:2097 ---> ACK sip:42@192.168.0.1;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.43:2063;branch=z9hG4bK-qceld7ydcw52;rport From: "43" ;tag=pj325n8gie To: ;tag=as77ba5dcc Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- asterisk*CLI> <--- SIP read from TLS://192.168.0.43:2097 ---> INVITE sip:42@192.168.0.1;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.43:2063;branch=z9hG4bK-lf2dwaymf9px;rport From: "43" ;tag=pj325n8gie To: Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="43",realm="asterisk-server",nonce="4aa19ad4",uri="sip:42@192.168.0.1;user=phone",response="4060b90d6ab0730aef2f3d56c31198f8",algorithm=MD5 Content-Type: application/sdp Content-Length: 475 v=0 o=root 1565129441 1565129441 IN IP4 192.168.0.43 s=call c=IN IP4 192.168.0.43 t=0 0 m=audio 57626 RTP/AVP 8 0 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Sg344vMy93smKq7f22kFiKjwavjW5TeBVUDAgJSG a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv �@�޷� l�޷ <-------------> --- (19 headers 20 lines) --- Sending to 192.168.0.43 : 2097 (NAT) Using INVITE request as basis request - 3c27c0545608-82f3mst3zcpd Found user '43' for '43' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.43:57626 Found audio description format pcma for ID 8 Found audio description format pcmu for ID 0 Found audio description format g722 for ID 9 Found audio description format g726-32 for ID 2 Found audio description format gsm for ID 3 Found audio description format g729 for ID 18 Found audio description format g723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=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 192.168.0.43:57626 Looking for 42 in fhvoip (domain 192.168.0.1) list_route: hop: <--- Transmitting (no NAT) to 192.168.0.43:2063 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.0.43:2063;branch=z9hG4bK-lf2dwaymf9px;received=192.168.0.43;rport=2097 From: "43" ;tag=pj325n8gie To: Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 2 INVITE User-Agent: FHVOIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [42@fhvoip:1] Answer("SIP/43-082722e8", "") in new stack Audio is at 192.168.0.1 port 13722 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk*CLI> <--- Reliably Transmitting (no NAT) to 192.168.0.43:2063 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.0.43:2063;branch=z9hG4bK-lf2dwaymf9px;received=192.168.0.43;rport=2097 From: "43" ;tag=pj325n8gie To: ;tag=as2077a122 Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 2 INVITE User-Agent: FHVOIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 529406687 529406687 IN IP4 192.168.0.1 s=Asterisk PBX 1.6.0-beta9 c=IN IP4 192.168.0.1 t=0 0 m=audio 13722 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 <------------> -- Executing [42@fhvoip:2] Dial("SIP/43-082722e8", "SIP/42") in new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.0.1 port 16470 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.42:2063: INVITE sip:42@192.168.0.42:2063;transport=tls;line=nuvzr06b SIP/2.0 Via: SIP/2.0/TLS 192.168.0.1:5060;branch=z9hG4bK4ce73f58;rport Max-Forwards: 70 From: "Snom" ;tag=as52d0dda6 To: Contact: Call-ID: 2b3aeb4667d9f8375d65924755df65ac@192.168.0.1 CSeq: 102 INVITE User-Agent: FHVOIP Date: Wed, 04 Jun 2008 08:02:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1217601730 1217601730 IN IP4 192.168.0.1 s=Asterisk PBX 1.6.0-beta9 c=IN IP4 192.168.0.1 t=0 0 m=audio 16470 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 42 asterisk*CLI> <--- SIP read from TLS://192.168.0.42:2096 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.0.1:5060;branch=z9hG4bK4ce73f58;rport=5061 From: "Snom" ;tag=as52d0dda6 To: ;tag=s0vwmkbxj1 Call-ID: 2b3aeb4667d9f8375d65924755df65ac@192.168.0.1 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/42-082738e0 is ringing asterisk*CLI> <--- SIP read from TLS://192.168.0.43:2097 ---> ACK sip:42@192.168.0.1:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 192.168.0.43:2063;branch=z9hG4bK-egtuyokum57t;rport From: "43" ;tag=pj325n8gie To: ;tag=as2077a122 Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.43, port 2063 Audio is at 192.168.0.1 port 13722 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.43:2063: INVITE sip:43@192.168.0.43:2063;transport=tls;line=se50cs9w SIP/2.0 Via: SIP/2.0/TLS 192.168.0.1:5060;branch=z9hG4bK7935f18a;rport Max-Forwards: 70 From: ;tag=as2077a122 To: "43" ;tag=pj325n8gie Contact: Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 102 INVITE User-Agent: FHVOIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 263 v=0 o=root 529406687 529406688 IN IP4 192.168.0.1 s=Asterisk PBX 1.6.0-beta9 c=IN IP4 192.168.0.1 t=0 0 m=audio 13722 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 --- asterisk*CLI> <--- SIP read from TLS://192.168.0.43:2097 ---> SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.1:5060;branch=z9hG4bK7935f18a;rport=5061 From: ;tag=as2077a122 To: "43" ;tag=pj325n8gie Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 327 v=0 o=root 1565129441 1565129442 IN IP4 192.168.0.43 s=call c=IN IP4 192.168.0.43 t=0 0 m=audio 57626 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Sg344vMy93smKq7f22kFiKjwavjW5TeBVUDAgJSG a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv 00 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv �@�޷� l�޷ <-------------> --- (13 headers 22 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.43:57626 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Found audio description format g729 for ID 18 Found audio description format g723 for ID 4 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x109 (g723|alaw|g729)/video=0x0 (nothing)/text=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 192.168.0.43:57626 --- set_address_from_contact host '192.168.0.43' set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.43, port 2063 Transmitting (no NAT) to 192.168.0.43:2063: ACK sip:43@192.168.0.43:2063;transport=tls;line=se50cs9w SIP/2.0 Via: SIP/2.0/TLS 192.168.0.1:5060;branch=z9hG4bK66b0efb7;rport Max-Forwards: 70 From: ;tag=as2077a122 To: "43" ;tag=pj325n8gie Contact: Call-ID: 3c27c0545608-82f3mst3zcpd CSeq: 102 ACK User-Agent: FHVOIP Content-Length: 0 --- -- Remote UNIX connection -- Remote UNIX connection disconnected asterisk*CLI> <--- SIP read from TLS://192.168.0.42:2096 ---> SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.1:5060;branch=z9hG4bK4ce73f58;rport=5061 From: "Snom" ;tag=as52d0dda6 To: ;tag=s0vwmkbxj1 Call-ID: 2b3aeb4667d9f8375d65924755df65ac@192.168.0.1 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom300/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 325 v=0 o=root 738243028 738243029 IN IP4 192.168.0.42 s=call c=IN IP4 192.168.0.42 t=0 0 m=audio 59736 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:QaZgjhX3nzz3/SXpMhJWG8y9viWRL80YVIucp7Rj a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (13 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.42:59736 Found audio description format pcma for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 192.168.0.42:59736 --- set_address_from_contact host '192.168.0.42' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.42, port 2063 Transmitting (no NAT) to 192.168.0.42:2063: ACK sip:42@192.168.0.42:2063;transport=tls;line=nuvzr06b SIP/2.0 Via: SIP/2.0/TLS 192.168.0.1:5060;branch=z9hG4bK431781a1;rport Max-Forwards: 70 From: "Snom" ;tag=as52d0dda6 To: ;tag=s0vwmkbxj1 Contact: Call-ID: 2b3aeb4667d9f8375d65924755df65ac@192.168.0.1 CSeq: 102 ACK User-Agent: FHVOIP Content-Length: 0 --- -- SIP/42-082738e0 answered SIP/43-082722e8 -- Native bridging SIP/43-082722e8 and SIP/42-082738e0 -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection Really destroying SIP dialog '3c26701b7c3c-m8mbuve9f95q' Method: REGISTER -- Remote UNIX connection disconnected Really destroying SIP dialog '3c27c03c4196-0hmg3r6ujeto' Method: REGISTER -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Reliably Transmitting (no NAT) to 192.168.0.44:5060: OPTIONS sip:44@192.168.0.44:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK15920d26;rport Max-Forwards: 70 From: "asterisk" ;tag=as734c5898 To: Contact: Call-ID: 34c086ae4232308b7ec196bd188c65e6@192.168.0.1 CSeq: 102 OPTIONS User-Agent: FHVOIP Date: Wed, 04 Jun 2008 08:03:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from UDP://192.168.0.44:5060 ---> SIP/2.0 200 OK To: ;tag=61b4e1f44d7a9dbci0 From: "asterisk" ;tag=as734c5898 Call-ID: 34c086ae4232308b7ec196bd188c65e6@192.168.0.1 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK15920d26 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '34c086ae4232308b7ec196bd188c65e6@192.168.0.1' Method: OPTIONS -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Reliably Transmitting (no NAT) to 192.168.0.42:2063: OPTIONS sip:42@192.168.0.42:2063;transport=tls;line=nuvzr06b SIP/2.0 Via: SIP/2.0/TLS 192.168.0.1:5060;branch=z9hG4bK07f04499;rport Max-Forwards: 70 From: "asterisk" ;tag=as6bc67ee8 To: Contact: Call-ID: 3aff2f7a2218b4b67face3cc494ddd4a@192.168.0.1 CSeq: 102 OPTIONS User-Agent: FHVOIP Date: Wed, 04 Jun 2008 08:03:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- asterisk*CLI> <--- SIP read from TLS://192.168.0.42:2096 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.0.1:5060;branch=z9hG4bK07f04499;rport=5061 From: "asterisk" ;tag=as6bc67ee8 To: Call-ID: 3aff2f7a2218b4b67face3cc494ddd4a@192.168.0.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom300/7.1.30 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '3aff2f7a2218b4b67face3cc494ddd4a@192.168.0.1' Method: OPTIONS asterisk*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message 192.168.0.42 42 2b3aeb4667d9f83 0x8 (alaw) No Tx: ACK 192.168.0.43 43 3c27c0545608-82 0x8 (alaw) No Tx: ACK 2 active SIP dialogs