<--- SIP read from UDP:192.168.1.172:16835 ---> INVITE sip:123456@192.168.1.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.172:16835;branch=z9hG4bKc7213db674 From: "ata-s1" ;tag=6f234fcc To: Call-ID: 58d2b953728525b3036e45823c955b90@192.168.1.172 Contact: CSeq: 1 INVITE Max-Forwards: 70 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK Allow-Events: talk,hold,conference Supported: replaces,100rel Content-Type: application/sdp User-Agent: CM5K-TA1S (807100.02) Content-Length: 302 v=0 o=CMI-SIPUA 10907 0 IN IP4 192.168.1.172 s=SIP CALL c=IN IP4 192.168.1.172 t=0 0 m=audio 20324 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 a=rtcp:20325 a=sendrecv <-------------> --- (14 headers 15 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.1.172 : 16835 (no NAT) Using INVITE request as basis request - 58d2b953728525b3036e45823c955b90@192.168.1.172 Found peer '1122334' for '1122334' from 192.168.1.172:16835 <--- Reliably Transmitting (NAT) to 192.168.1.172:16835 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.172:16835;branch=z9hG4bKc7213db674;received=192.168.1.172 From: "ata-s1" ;tag=6f234fcc To: ;tag=as5b80edee Call-ID: 58d2b953728525b3036e45823c955b90@192.168.1.172 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="492b2b2c" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '58d2b953728525b3036e45823c955b90@192.168.1.172' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.1.172:16835 ---> ACK sip:123456@192.168.1.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.172:16835;branch=z9hG4bKc7213db674 From: "ata-s1" ;tag=6f234fcc To: ;tag=as5b80edee Call-ID: 58d2b953728525b3036e45823c955b90@192.168.1.172 Contact: CSeq: 1 ACK Max-Forwards: 70 User-Agent: CM5K-TA1S (807100.02) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.1.172:16835 ---> INVITE sip:123456@192.168.1.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.172:16835;branch=z9hG4bK64b8ef2376 From: "ata-s1" ;tag=6f234fcc To: Call-ID: 58d2b953728525b3036e45823c955b90@192.168.1.172 Contact: CSeq: 2 INVITE Max-Forwards: 70 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK Allow-Events: talk,hold,conference Authorization: Digest username="1122334",realm="asterisk",nonce="492b2b2c",response="8ab1b12888aa15e0e5d0bc908626dc56",uri="sip:123456@192.168.1.110",algorithm=MD5 Supported: replaces,100rel Content-Type: application/sdp User-Agent: CM5K-TA1S (807100.02) Content-Length: 302 v=0 o=CMI-SIPUA 10907 0 IN IP4 192.168.1.172 s=SIP CALL c=IN IP4 192.168.1.172 t=0 0 m=audio 20324 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 a=rtcp:20325 a=sendrecv <-------------> --- (15 headers 15 lines) --- Sending to 192.168.1.172 : 16835 (NAT) Using INVITE request as basis request - 58d2b953728525b3036e45823c955b90@192.168.1.172 Found peer '1122334' for '1122334' from 192.168.1.172:16835 Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G729 for ID 18 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 - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.172:20324 Looking for 123456 in outcontext (domain 192.168.1.110) list_route: hop: <--- Transmitting (NAT) to 192.168.1.172:16835 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.172:16835;branch=z9hG4bK64b8ef2376;received=192.168.1.172 From: "ata-s1" ;tag=6f234fcc To: Call-ID: 58d2b953728525b3036e45823c955b90@192.168.1.172 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [123456@outcontext:1] Answer("SIP/1122334-00000000", "") in new stack Audio is at 192.168.1.110 port 15712 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.1.172:16835 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.172:16835;branch=z9hG4bK64b8ef2376;received=192.168.1.172 From: "ata-s1" ;tag=6f234fcc To: ;tag=as1db6eec3 Call-ID: 58d2b953728525b3036e45823c955b90@192.168.1.172 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.17 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 1120191583 1120191583 IN IP4 192.168.1.110 s=Asterisk PBX 1.6.2.17 c=IN IP4 192.168.1.110 t=0 0 m=audio 15712 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.1.172:16835 ---> ACK sip:123456@192.168.1.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.172:16835;branch=z9hG4bK3babcf7cf3 From: "ata-s1" ;tag=6f234fcc To: ;tag=as1db6eec3 Call-ID: 58d2b953728525b3036e45823c955b90@192.168.1.172 Contact: CSeq: 2 ACK Max-Forwards: 70 Authorization: Digest username="1122334",realm="asterisk",nonce="492b2b2c",response="a5fb3ac8eead2503e2602e76d9ca34b5",uri="sip:123456@192.168.1.110",algorithm=MD5 User-Agent: CM5K-TA1S (807100.02) Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- Executing [123456@outcontext:2] Playback("SIP/1122334-00000000", "hello-world") in new stack -- Playing 'hello-world.gsm' (language 'en') -- Executing [123456@outcontext:3] Hangup("SIP/1122334-00000000", "") in new stack == Spawn extension (outcontext, 123456, 3) exited non-zero on 'SIP/1122334-00000000' Scheduling destruction of SIP dialog '58d2b953728525b3036e45823c955b90@192.168.1.172' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.172, port 16835 Reliably Transmitting (NAT) to 192.168.1.172:16835: BYE sip:1122334@192.168.1.172:16835 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bK18f78653;rport Max-Forwards: 70 From: ;tag=as1db6eec3 To: "ata-s1" ;tag=6f234fcc Call-ID: 58d2b953728525b3036e45823c955b90@192.168.1.172 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.2.17 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:192.168.1.172:16835 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.110:5060;rport=5060;received=192.168.1.110;branch=z9hG4bK18f78653 From: ;tag=as1db6eec3 To: "ata-s1" ;tag=6f234fcc Call-ID: 58d2b953728525b3036e45823c955b90@192.168.1.172 CSeq: 102 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) ---