<-- SIP read from 212.59.4.20:5060: INVITE sip:370XXXXXXX@192.168.1.104:5061 SIP/2.0 Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08njl830aor0eagre640.1 To: From: ;tag=test_tag_0009976054 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 1235 INVITE Contact: Accept-Language: en;q=0.0 Alert-Info: Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Wed,30 May 2007 18:02:18 GMT Max-Forwards: 68 Supported: timer Session-Expires: 1800;refresher=uac Min-SE: 1800 Content-Type: application/sdp Content-Length: 626 v=0 o=hiQ9200 4476420070430210311 1785724934 IN IP4 212.59.4.20 s=Phone Call via hiQ9200 SIPCA c=IN IP4 212.59.4.20 t=0 0 m=audio 15284 RTP/AVP 8 18 0 4 102 2 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:2 G726-32/8000 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 18 0 4 102 2 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=fmtp:8 vad=no a=cpar: a=rtpmap:18 G729/8000 a=cpar: a=rtpmap:0 PCMU/8000 a=cpar: a=rtpmap:4 G723/8000 a=cpar: a=rtpmap:102 G726-32/8000 a=cpar: a=rtpmap:2 G726-32/8000 a=cdsc: 7 image udptl t38 a=sendrecv --- (17 headers 24 lines) --- Using INVITE request as basis request - 8318450811-7267935675490811-11-863266 Sending to 212.59.4.20 : 5060 (non-NAT) Found peer 'teo' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 102 Found RTP audio format 2 Peer audio RTP is at port 212.59.4.20:15284 Found description format PCMA Found description format G729 Found description format PCMU Found description format G723 Found description format G726-32 Found description format G726-32 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x11d (g723|ulaw|alaw|g726|g729 )/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 370XXXXXXX in default (domain 192.168.1.104) list_route: hop: Transmitting (NAT) to 212.59.4.20:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08njl830aor0eagre640.1;received= 212.59.4.20 From: ;tag=test_tag_0009976054 To: Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 1235 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/370XXXXXXX-08c85580", "SIP/602") in new stack We're at 192.168.1.104 port 18022 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.1.113:5060: INVITE sip:602@192.168.1.113:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK7b448697;rport From: "0037067198101" ;tag=as49e8ab17 To: Contact: Call-ID: 303d40b879931d1e32c3847b0c56c7d9@192.168.1.104 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 30 May 2007 18:00:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 8656 8656 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 18022 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 602 <-- SIP read from 192.168.1.113:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK7b448697;rport From: "0037067198101" ;tag=as49e8ab17 To: Call-ID: 303d40b879931d1e32c3847b0c56c7d9@192.168.1.104 CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines) --- <-- SIP read from 192.168.1.113:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK7b448697;rport From: "0037067198101" ;tag=as49e8ab17 To: ;tag=2320332316 Call-ID: 303d40b879931d1e32c3847b0c56c7d9@192.168.1.104 CSeq: 102 INVITE Contact: Content-Length: 0 --- (8 headers 0 lines) --- -- SIP/602-08c8aeb0 is ringing Transmitting (NAT) to 212.59.4.20:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08njl830aor0eagre640.1;received= 212.59.4.20 From: ;tag=test_tag_0009976054 To: ;tag=as7db60021 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 1235 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Destroying call '1051226431-316095009@192.168.1.113' <-- SIP read from 192.168.1.113:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK7b448697;rport From: "0037067198101" ;tag=as49e8ab17 To: ;tag=2320332316 Call-ID: 303d40b879931d1e32c3847b0c56c7d9@192.168.1.104 CSeq: 102 INVITE Contact: supported: replaces Content-Type: application/sdp Content-Length: 171 v=0 o=606 15826124 14458959 IN IP4 192.168.1.113 s=A conversation c=IN IP4 192.168.1.113 t=0 0 m=audio 10172 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 --- (10 headers 8 lines) --- Found RTP audio format 0 Found RTP audio format 8 Peer audio RTP is at port 192.168.1.113:10172 Found description format PCMU Found description format PCMA Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/vi deo=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combin ed - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send t o set_destination: set destination to 192.168.1.113, port 5060 Transmitting (no NAT) to 192.168.1.113:5060: ACK sip:602@192.168.1.113:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK7f58c70e;rport From: "0037067198101" ;tag=as49e8ab17 To: ;tag=2320332316 Contact: Call-ID: 303d40b879931d1e32c3847b0c56c7d9@192.168.1.104 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/602-08c8aeb0 answered SIP/370XXXXXXX-08c85580 We're at 192.168.1.104 port 13776 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (NAT) to 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08njl830aor0eagre640.1;received= 212.59.4.20 From: ;tag=test_tag_0009976054 To: ;tag=as7db60021 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 1235 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 184 v=0 o=root 8656 8656 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 13776 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/370XXXXXXX-08c85580 and SIP/602-08c8aeb0 <-- SIP read from 212.59.4.20:5060: ACK sip:370XXXXXXX@192.168.1.104:5061 SIP/2.0 Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08nj5c30bghgbagr3701.1 To: ;tag=as7db60021 From: ;tag=test_tag_0009976054 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 1235 ACK Contact: Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Wed,30 May 2007 18:02:25 GMT Max-Forwards: 68 Content-Length: 0 --- (11 headers 0 lines) --- <-- SIP read from 212.59.4.20:5060: INVITE sip:370XXXXXXX@192.168.1.104:5061 SIP/2.0 Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08nj5c30b8vg6aks1201.1 To: ;tag=as7db60021 From: ;tag=test_tag_0009976054 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 1236 INVITE Contact: Accept-Language: en;q=0.0 Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Wed,30 May 2007 18:02:25 GMT Max-Forwards: 68 Supported: timer Session-Expires: 1800;refresher=uac Min-SE: 1800 Content-Type: application/sdp Content-Length: 186 v=0 o=hiQ9200 4476420070430210311 1785724935 IN IP4 212.59.4.20 s=Phone Call via hiQ9200 SIPCA c=IN IP4 212.59.4.20 t=0 0 m=audio 15284 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 --- (16 headers 8 lines) --- Using INVITE request as basis request - 8318450811-7267935675490811-11-863266 Sending to 212.59.4.20 : 5060 (NAT) Found RTP audio format 0 Peer audio RTP is at port 212.59.4.20:15284 Found description format PCMU Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 192.168.1.104 port 13776 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08nj5c30b8vg6aks1201.1;received= 212.59.4.20 From: ;tag=test_tag_0009976054 To: ;tag=as7db60021 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 1236 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 160 v=0 o=root 8656 8657 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 13776 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- <-- SIP read from 212.59.4.20:5060: ACK sip:370XXXXXXX@192.168.1.104:5061 SIP/2.0 Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08nj5c30copgca0vf2k1.1 To: ;tag=as7db60021 From: ;tag=test_tag_0009976054 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 1236 ACK Contact: Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Wed,30 May 2007 18:02:25 GMT Max-Forwards: 68 Content-Length: 0 --- (11 headers 0 lines) --- May 30 21:00:50 NOTICE[8676]: rtp.c:331 process_rfc3389: Comfort noise support i ncomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 212.59.4.20 May 30 21:00:52 NOTICE[8668]: chan_sip.c:5473 sip_reregister: -- Re-registrat ion for 370XXXXXXX@skambink.zebra.lt REGISTER 13 headers, 0 lines REGISTER attempt 1 to 370XXXXXXX@skambink.zebra.lt Reliably Transmitting (no NAT) to 212.59.4.20:5060: REGISTER sip:skambink.zebra.lt SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK48cdf2c3;rport From: ;tag=as604c3ea0 To: Call-ID: 5b7706dc4ad77dac6133a6ed35f39985@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="370XXXXXXX", realm="voip.telecom.lt", algorithm =MD5, uri="sip:skambink.zebra.lt", nonce="3215546643-6e577cf695fcada545b867100e5 d96e7", response="007c9c65dbcf316c5a18a94c9340dc59", opaque="", qop=auth, cnonce ="107fa6c4", nc=00000002 Expires: 120 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5061;received=82.135.246.157;branch=z9hG4bK48cdf2 c3;rport=5061 From: ;tag=as604c3ea0 To: ;tag=aprqfbbc7i1-4httnk10000g6 Call-ID: 5b7706dc4ad77dac6133a6ed35f39985@127.0.0.1 CSeq: 104 REGISTER Contact: ;expires=30 --- (7 headers 0 lines) --- Scheduling destruction of call '5b7706dc4ad77dac6133a6ed35f39985@127.0.0.1' in 3 2000 ms May 30 21:00:52 NOTICE[8668]: chan_sip.c:10009 handle_response_register: Outboun d Registration: Expiry for skambink.zebra.lt is 30 sec (Scheduling reregistratio n in 23 s) <-- SIP read from 192.168.1.113:5060: BYE sip:0037067198101@192.168.1.104:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK12971690395631045 From: ;tag=2320332316 To: "0037067198101" ;tag=as49e8ab17 Call-ID: 303d40b879931d1e32c3847b0c56c7d9@192.168.1.104 CSeq: 1 BYE max-forwards: 70 user-agent: Voip Phone 1.0 Content-Length: 0 --- (9 headers 0 lines) --- Sending to 192.168.1.113 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.1.113:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK12971690395631045;received=192 .168.1.113 From: ;tag=2320332316 To: "0037067198101" ;tag=as49e8ab17 Call-ID: 303d40b879931d1e32c3847b0c56c7d9@192.168.1.104 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- == Spawn extension (default, 370XXXXXXX, 1) exited non-zero on 'SIP/370521999 62-08c85580' Scheduling destruction of call '8318450811-7267935675490811-11-863266' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 212.59.4.20, port 5060 Reliably Transmitting (NAT) to 212.59.4.20:5060: BYE sip:0037067198101@212.59.4.20:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK668bc8b0;rport From: ;tag=as7db60021 To: ;tag=test_tag_0009976054 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <-- SIP read from 212.59.4.20:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.104:5061;received=82.135.246.157;branch=z9hG4bK668bc8 b0;rport=5061 From: ;tag=as7db60021 To: ;tag=test_tag_0009976054 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 WWW-Authenticate: Digest realm="voip.telecom.lt", nonce="3215546931-3513c291abb8 677ff4e413a2962cd389", algorithm=MD5, qop="auth" --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 212.59.4.20, port 5060 Reliably Transmitting (NAT) to 212.59.4.20:5060: BYE sip:0037067198101@212.59.4.20:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK2cbc496e;rport From: ;tag=as7db60021 To: ;tag=test_tag_0009976054 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="370XXXXXXX", realm="voip.telecom.lt", algorithm =MD5, uri="sip:0037067198101@212.59.4.20:5060", nonce="3215546931-3513c291abb867 7ff4e413a2962cd389", response="8e8c0d094476be20cefc8a81d1466a2c", opaque="", qop =auth, cnonce="2a1355b2", nc=00000001 Date: Wed, 30 May 2007 18:00:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '303d40b879931d1e32c3847b0c56c7d9@192.168.1.104' <-- SIP read from 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5061;received=82.135.246.157;branch=z9hG4bK2cbc49 6e;rport=5061 From: ;tag=as7db60021 To: ;tag=test_tag_0009976054 Call-ID: 8318450811-7267935675490811-11-863266 CSeq: 103 BYE Contact: Date: Wed,30 May 2007 18:02:32 GMT Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '8318450811-7267935675490811-11-863266' Destroying call '5b7706dc4ad77dac6133a6ed35f39985@127.0.0.1'