<--- SIP read from UDP:10.0.20.12:5060 ---> INVITE sip:2104587045@10.0.20.45 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.12:5060;branch=z9hG4bK1306485535;rport From: ;tag=666918886 To: Call-ID: 574340173-5060-7@BA.A.CA.BC CSeq: 50 INVITE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2100 1.0.1.56 Privacy: none P-Preferred-Identity: Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 372 v=0 o=12 8000 8000 IN IP4 10.0.20.12 s=SIP Call c=IN IP4 10.0.20.12 t=0 0 m=audio 5004 RTP/AVP 8 97 4 18 9 2 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (17 headers 18 lines) --- Sending to 10.0.20.12:5060 (no NAT) Using INVITE request as basis request - 574340173-5060-7@BA.A.CA.BC Found peer '12' for '12' from 10.0.20.12:5060 <--- Reliably Transmitting (no NAT) to 10.0.20.12:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.20.12:5060;branch=z9hG4bK1306485535;received=10.0.20.12;rport=5060 From: ;tag=666918886 To: ;tag=as42a03233 Call-ID: 574340173-5060-7@BA.A.CA.BC CSeq: 50 INVITE Server: Asterisk PBX 10.0.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="495a3ff2" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '574340173-5060-7@BA.A.CA.BC' in 32000 ms (Method: INVITE) <--- SIP read from UDP:10.0.20.12:5060 ---> ACK sip:2104587045@10.0.20.45 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.12:5060;branch=z9hG4bK1306485535;rport From: ;tag=666918886 To: ;tag=as42a03233 Call-ID: 574340173-5060-7@BA.A.CA.BC CSeq: 50 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:10.0.20.12:5060 ---> INVITE sip:2104587045@10.0.20.45 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.12:5060;branch=z9hG4bK277747968;rport From: ;tag=666918886 To: Call-ID: 574340173-5060-7@BA.A.CA.BC CSeq: 51 INVITE Contact: Authorization: Digest username="12", realm="asterisk", nonce="495a3ff2", uri="sip:2104587045@10.0.20.45", response="bf87e970c6189911343e017bed8dc2fb", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2100 1.0.1.56 Privacy: none P-Preferred-Identity: Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 372 v=0 o=12 8000 8000 IN IP4 10.0.20.12 s=SIP Call c=IN IP4 10.0.20.12 t=0 0 m=audio 5004 RTP/AVP 8 97 4 18 9 2 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (18 headers 18 lines) --- Sending to 10.0.20.12:5060 (no NAT) Using INVITE request as basis request - 574340173-5060-7@BA.A.CA.BC Found peer '12' for '12' from 10.0.20.12:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio description format G723 for ID 4 Found audio description format G729 for ID 18 Found audio description format G722 for ID 9 Found audio description format G726-32 for ID 2 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(g723|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.0.20.12:5004 Looking for 2104587045 in users (domain 10.0.20.45) list_route: hop: <--- Transmitting (no NAT) to 10.0.20.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.20.12:5060;branch=z9hG4bK277747968;received=10.0.20.12;rport=5060 From: ;tag=666918886 To: Call-ID: 574340173-5060-7@BA.A.CA.BC CSeq: 51 INVITE Server: Asterisk PBX 10.0.0-beta1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [2104587045@users:1] Dial("SIP/12-0000001a", "sip/trunk/2104587045") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 100018 (silk8) to SDP Adding codec 100018 (silk12) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.88.105:5060: INVITE sip:2104587045@192.168.88.105 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.127:5060;branch=z9hG4bK02aefc0c Max-Forwards: 70 From: "12" ;tag=as5f1e9293 To: Contact: Call-ID: 4f12a21a5dcb4b732b06efc93cd3a392@192.168.88.127:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 10.0.0-beta1 Date: Sun, 25 Sep 2011 15:26:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 468 v=0 o=root 1967040386 1967040386 IN IP4 192.168.88.127 s=Asterisk PBX 10.0.0-beta1 c=IN IP4 192.168.88.127 t=0 0 m=audio 10376 RTP/AVP 96 100 101 a=rtpmap:96 SILK/8000 a=fmtp:96 maxaveragebitrate=10000 a=fmtp:96 usedtx=1 a=fmtp:96 useinbandfec=1 a=rtpmap:100 SILK/12000 a=fmtp:100 maxaveragebitrate=12000 a=fmtp:100 usedtx=0 a=fmtp:100 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---