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#247739 ·published 2006-11-13 20:42 UTC
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Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:15742042820@208.100.6.10>
Proxy-Authenticate: Digest realm="asterisk", nonce="46bb78ef"
Content-Length: 0


Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 0 lines)Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 0 lines)---
Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Acked pending invite 102
Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Stopping retransmission on '372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177' of Request 102: Match Found
Nov 13 14:38:04 VERBOSE[8868] logger.c: Transmitting (no NAT) to 208.100.6.10:5060:UDP
 with fd 12
ACK sip:15742042820@chiv1.voipstreet.com SIP/2.0
Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK0bc67443;rport
From: "OnsiteServiceNow" <sip:5742064786@70.90.149.177>;tag=as24196c6e
To: <sip:15742042820@chiv1.voipstreet.com>;tag=as58470746
Contact: <sip:5742064786@70.90.149.177;transport=UDP>
Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Nov 13 14:38:04 VERBOSE[8868] logger.c: UDP write: fd 12
Nov 13 14:38:04 VERBOSE[8868] logger.c: We're at 70.90.149.177 port 16668
Nov 13 14:38:04 VERBOSE[8868] logger.c: Adding codec 0x4 (ulaw) to SDP
Nov 13 14:38:04 VERBOSE[8868] logger.c: Adding codec 0x8 (alaw) to SDP
Nov 13 14:38:04 VERBOSE[8868] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Nov 13 14:38:04 VERBOSE[8868] logger.c: Reliably Transmitting (no NAT) to 208.100.6.10:5060:UDP
 with fd 12
INVITE sip:15742042820@chiv1.voipstreet.com SIP/2.0
Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK69971141;rport
From: "OnsiteServiceNow" <sip:5742064786@70.90.149.177>;tag=as24196c6e
To: <sip:15742042820@chiv1.voipstreet.com>
Contact: <sip:5742064786@70.90.149.177;transport=UDP>
Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="dfrohwerk-17891", realm="asterisk", algorithm=MD5, uri="sip:15742042820@chiv1.voipstreet.com", nonce="46bb78ef", response="282a4f2840c689b13f6a71f5c54a64f5", opaque=""
Date: Mon, 13 Nov 2006 20:38:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 8858 8859 IN IP4 70.90.149.177
s=session
c=IN IP4 70.90.149.177
t=0 0
m=audio 16668 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=silenceSupp:off - - - -

---
Nov 13 14:38:04 VERBOSE[8868] logger.c: UDP write: fd 12
Nov 13 14:38:04 VERBOSE[8868] logger.c:
<-- Sip read from 208.100.6.10:5060:UDP
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK69971141;received=70.90.149.177;rport=5060
From: "OnsiteServiceNow" <sip:5742064786@70.90.149.177>;tag=as24196c6e
To: <sip:15742042820@chiv1.voipstreet.com>
Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177
CSeq: 103 INVITE
User-Agent: Lightwave
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:15742042820@208.100.6.10>
Content-Length: 0


Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (10 headers 0 lines)Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (10 headers 0 lines)---
Nov 13 14:38:04 DEBUG[8868] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177' Request 103: Found
Nov 13 14:38:04 VERBOSE[8868] logger.c:
<-- Sip read from 208.100.6.10:5060:UDP
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK69971141;received=70.90.149.177;rport=5060
From: "OnsiteServiceNow" <sip:5742064786@70.90.149.177>;tag=as24196c6e
To: <sip:15742042820@chiv1.voipstreet.com>;tag=as33dba6e8
Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177
CSeq: 103 INVITE
User-Agent: Lightwave
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:15742042820@208.100.6.10>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 4534 4534 IN IP4 208.100.6.10
s=session
c=IN IP4 208.100.6.10
t=0 0
m=audio 14226 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 10 lines)Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 10 lines)---
Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Acked pending invite 103
Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Stopping retransmission on '372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177' of Request 103: Match Found
Nov 13 14:38:04 VERBOSE[8868] logger.c: Found RTP audio format 0
Nov 13 14:38:04 VERBOSE[8868] logger.c: Found RTP audio format 101
Nov 13 14:38:04 VERBOSE[8868] logger.c: Peer audio RTP is at port 208.100.6.10:14226
Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Peer audio RTP is at port 208.100.6.10:14226
Nov 13 14:38:04 VERBOSE[8868] logger.c: Found description format PCMU
Nov 13 14:38:04 VERBOSE[8868] logger.c: Found description format telephone-event
Nov 13 14:38:04 VERBOSE[8868] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Nov 13 14:38:04 VERBOSE[8868] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Nov 13 14:38:04 DEBUG[8868] chan_sip.c: build_route: Contact hop: <sip:15742042820@208.100.6.10>
Nov 13 14:38:04 VERBOSE[8868] logger.c: list_route: hop: <sip:15742042820@208.100.6.10>
Nov 13 14:38:04 VERBOSE[8868] logger.c: set_destination: Parsing <sip:15742042820@208.100.6.10> for address/port/transport to send to
Nov 13 14:38:04 WARNING[8868] chan_sip.c: Unsupported transport 'UDP' should be 'UDP', 'TCP', or 'TLS'
Nov 13 14:38:04 VERBOSE[8868] logger.c: Transmitting (no NAT) to 208.100.6.10:5060:UDP
 with fd 12
ACK sip:15742042820@208.100.6.10 SIP/2.0
Via: SIP/2.0/UDP 70.90.149.177:5060;branch=z9hG4bK3f4f13b0;rport
From: "OnsiteServiceNow" <sip:5742064786@70.90.149.177>;tag=as24196c6e
To: <sip:15742042820@chiv1.voipstreet.com>;tag=as33dba6e8
Contact: <sip:5742064786@70.90.149.177;transport=UDP>
Call-ID: 372385330dcc4f5d6bba8e5b34dd95d9@70.90.149.177
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Nov 13 14:38:04 VERBOSE[8868] logger.c: UDP write: fd 12
Nov 13 14:38:04 VERBOSE[10005] logger.c:     -- SIP/voipstreet-082b98f0 answered SIP/201-08294c38
Nov 13 14:38:04 VERBOSE[10005] logger.c: We're at 70.90.149.177 port 10812
Nov 13 14:38:04 VERBOSE[10005] logger.c: Adding codec 0x4 (ulaw) to SDP
Nov 13 14:38:04 VERBOSE[10005] logger.c: Adding codec 0x8 (alaw) to SDP
Nov 13 14:38:04 VERBOSE[10005] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Nov 13 14:38:04 VERBOSE[10005] logger.c: Reliably Transmitting (no NAT) to 70.90.149.179:5060:UDP
 with fd 12
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.90.149.179:5060;branch=z9hG4bK-f003f7d8;received=70.90.149.179
From: Onsite Service Now,Inc. <sip:201@70.90.149.177>;tag=4a2a0c47cf2fb0aco0
To: <sip:15742042820@70.90.149.177>;tag=as60d218e5
Call-ID: 33c0cca3-6328c00@70.90.149.179
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:15742042820@70.90.149.177;transport=UDP>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 8858 8858 IN IP4 70.90.149.177
s=session
c=IN IP4 70.90.149.177
t=0 0
m=audio 10812 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=silenceSupp:off - - - -

---
Nov 13 14:38:04 VERBOSE[10005] logger.c: UDP write: fd 12
Nov 13 14:38:04 VERBOSE[10005] logger.c:     -- Attempting native bridge of SIP/201-08294c38 and SIP/voipstreet-082b98f0
Nov 13 14:38:04 VERBOSE[8868] logger.c:
<-- Sip read from 70.90.149.179:5060:UDP
ACK sip:15742042820@70.90.149.177;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 70.90.149.179:5060;branch=z9hG4bK-53d7d29d
From: Onsite Service Now,Inc. <sip:201@70.90.149.177>;tag=4a2a0c47cf2fb0aco0
To: <sip:15742042820@70.90.149.177>;tag=as60d218e5
Call-ID: 33c0cca3-6328c00@70.90.149.179
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="201",realm="asterisk",nonce="035c93d5",uri="sip:15742042820@70.90.149.177",algorithm=MD5,response="ff81d0af59a1914336f4aca9c774dce9"
Contact: Onsite Service Now,Inc. <sip:201@70.90.149.179:5060>
User-Agent: Sipura/SPA1000-2.0.10(e)
Content-Length: 0


Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 0 lines)Nov 13 14:38:04 VERBOSE[8868] logger.c: --- (11 headers 0 lines)---
Nov 13 14:38:04 DEBUG[8868] chan_sip.c: Stopping retransmission on '33c0cca3-6328c00@70.90.149.179' of Response 102: Match Found
Nov 13 14:38:06 VERBOSE[8868] logger.c:
<-- Sip read from 70.90.149.179:5060:UDP
NOTIFY sip:70.90.149.177 SIP/2.0
Via: SIP/2.0/UDP 70.90.149.179:5060;branch=z9hG4bK-fc91af68
From: Onsite Service Now,Inc. <sip:201@70.90.149.177>;tag=9d074f5b40d37cfo0
To: <sip:70.90.149.177>
Call-ID: 43d8809b-666bb78f@70.90.149.179
CSeq: 933 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA1000-2.0.10(e)
Content-Length: 0


Nov 13 14:38:06 VERBOSE[8868] logger.c: --- (10 headers 0 lines)Nov 13 14:38:06 VERBOSE[8868] logger.c: --- (10 headers 0 lines)---
Nov 13 14:38:06 VERBOSE[8868] logger.c: Transmitting (no NAT) to 70.90.149.179:5060:UDP
 with fd 12
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.90.149.179:5060;branch=z9hG4bK-fc91af68;received=70.90.149.179
From: Onsite Service Now,Inc. <sip:201@70.90.149.177>;tag=9d074f5b40d37cfo0
To: <sip:70.90.149.177>;tag=as34aec288
Call-ID: 43d8809b-666bb78f@70.90.149.179
CSeq: 933 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Nov 13 14:38:06 VERBOSE[8868] logger.c: UDP write: fd 12
Nov 13 14:38:06 VERBOSE[8868] logger.c: Destroying call '43d8809b-666bb78f@70.90.149.179'

[2]+  Stopped                 tail -f /var/log/asterisk/full