[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 0: INVITE sip:036108YYYY@202.52.129.50:5060;user=phone SIP/2.0 (59)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 1: Via: SIP/2.0/UDP 125.213.160.7:5060;branch=z9hG4bK17dd5a0074cb31ee2-cd209-0 (75)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 2: Max-Forwards: 70 (16)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 3: Contact: <sip:041772XXXX@125.213.160.7:5060> (44)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 4: To: <sip:036108YYYY@202.52.129.50:5060> (39)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 5: From: "041772XXXX"<sip:041772XXXX@125.213.160.7:5060>;tag=5c65bd79-co8402-INS001 (80)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 6: Call-ID: 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3 (58)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 7: CSeq: 840201 INVITE (19)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 8: Content-Type: application/sdp (29)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 9: Supported: 100rel (17)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 10: User-Agent: ENSR2.5.4 (21)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 11: Content-Length: 452 (19)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 12:  (0)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: v=0 (3)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: o=- 1550171513 1550171513 IN IP4 125.213.160.7 (46)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: s=ENSResip (10)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: c=IN IP4 125.213.160.11 (23)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: t=0 0 (5)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: m=audio 17090 RTP/AVP 18 2 98 8 0 97 4 101 (42)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=fmtp:18 annexb=yes (20)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=fmtp:98 mode=20 (17)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=fmtp:97 mode=30 (17)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=fmtp:101 0-15 (15)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=rtpmap:2 G726-32/8000 (23)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=rtpmap:98 iLBC/8000 (21)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=rtpmap:97 iLBC/8000 (21)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=silenceSupp:off - - - - (25)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Line: a=sendrecv (10)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Setting NAT on RTP to Off
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Setting NAT on UDPTL to Off
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Allocating new SIP dialog for 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3 - INVITE (With RTP)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Begin: parsing SIP "Supported: 100rel"
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Found SIP option: -100rel-
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Matched SIP option: 100rel
[2010-10-12 00:27:46] DEBUG[21634] res_config_mysql.c: MySQL RealTime: Everything is fine.
[2010-10-12 00:27:46] DEBUG[21634] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_accounts WHERE name = '041772XXXX'
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Setting NAT on RTP to Off
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Setting NAT on UDPTL to Off
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing session-level SDP o=- 1550171513 1550171513 IN IP4 125.213.160.7... UNSUPPORTED.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing session-level SDP s=ENSResip... UNSUPPORTED.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing session-level SDP c=IN IP4 125.213.160.11... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=fmtp:98 mode=20... UNSUPPORTED.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=fmtp:97 mode=30... UNSUPPORTED.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 iLBC/8000... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 iLBC/8000... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:4 G723/8000... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: T38 state changed to 0 on channel <none>
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: We're settling with these formats: 0x8 (alaw)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Checking SIP call limits for device 
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Updating call counter for incoming call
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: *** Our native formats are 0x8 (alaw) 
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: *** Joint capabilities are 0x8 (alaw) 
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: *** Our capabilities are 0x8 (alaw) 
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) 
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: This channel will not be able to handle video.
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: build_route: Contact hop: <sip:041772XXXX@125.213.160.7:5060>
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: SIP/fromsymbio-00000043: New call is still down.... Trying... 
[2010-10-12 00:27:46] DEBUG[21634] devicestate.c: Notification of state change to be queued on device/channel SIP/fromsymbio
[2010-10-12 00:27:46] DEBUG[21631] devicestate.c: No provider found, checking channel drivers for SIP - fromsymbio
[2010-10-12 00:27:46] DEBUG[21631] chan_sip.c: Checking device state for peer fromsymbio
[2010-10-12 00:27:46] DEBUG[21631] devicestate.c: Changing state for SIP/fromsymbio - state 1 (Not in use)
[2010-10-12 00:27:46] DEBUG[21651] app_queue.c: Device 'SIP/fromsymbio' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'Set'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'Set'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'Goto'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'Set'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'Set'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'Set'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'Progress'
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Setting framing from config on incoming call
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: -- Done with adding codecs to SDP
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw)
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'PlayTones'
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Set channel SIP/fromsymbio-00000043 to write format slin
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Prodding channel 'SIP/fromsymbio-00000043'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Function result is '041772XXXX'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Function result is '041772XXXX'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Function result is '<sip:041772XXXX@125.213.160.7:5060>'
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'AGI'
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Invalid SIP message - rejected , no callid, len 382
[2010-10-12 00:27:46] ERROR[23038] utils.c: write() returned error: Broken pipe
[2010-10-12 00:27:46] DEBUG[23038] pbx.c: Launching 'Dial'
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Our T38 capability (3872)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Setting NAT on RTP to On
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Setting NAT on UDPTL to On
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Destroying SIP peer 74189
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: *** Our native formats are 0x4 (ulaw) 
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: *** Joint capabilities are 0x0 (nothing) 
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: *** Our capabilities are 0x10c (ulaw|alaw|g729) 
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) 
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) 
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: This channel will not be able to handle video.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable DIALEDTIME.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable ANSWEREDTIME.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable DIALEDPEERNAME.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable DIALEDPEERNUMBER.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable DIALSTATUS.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable AGISTATUS.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable TO_DIAL2.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable TO_DIAL.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable TRUNK_GROUP.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable TRUNK_ORDER.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable DIVERT_DEST.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable PASSERT.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable CLIENTID.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable DIDNO.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable RING_TIME.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable CALLING_NUMBER.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable TRUECID.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable RELATED_CHANNELS.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable FROM_DID.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable EXTN_TO_DIAL.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable SIPCALLID.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable SIPUSERAGENT.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable SIPDOMAIN.
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Not copying variable SIPURI.
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Outgoing Call for 036108YYYY
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Updating call counter for outgoing call
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Our T38 capability (3872), joint T38 capability (3872)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: ** Our capability: 0x10c (ulaw|alaw|g729) Video flag: False
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: ** Our prefcodec: 0x8 (alaw) 
[2010-10-12 00:27:46] VERBOSE[23038] logger.c: Audio is at 202.52.129.50 port 18216
[2010-10-12 00:27:46] VERBOSE[23038] logger.c: Adding codec 0x8 (alaw) to SDP
[2010-10-12 00:27:46] VERBOSE[23038] logger.c: Adding codec 0x4 (ulaw) to SDP
[2010-10-12 00:27:46] VERBOSE[23038] logger.c: Adding codec 0x100 (g729) to SDP
[2010-10-12 00:27:46] VERBOSE[23038] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: -- Done with adding codecs to SDP
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Done building SDP. Settling with this capability: 0x10c (ulaw|alaw|g729)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 0: INVITE sip:036108YYYY@27.50.95.170:5060 SIP/2.0 (47)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 1: Via: SIP/2.0/UDP 202.52.129.50:5060;branch=z9hG4bK7afce819;rport (64)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 2: From: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6 (64)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 3: To: <sip:036108YYYY@27.50.95.170:5060> (38)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 4: Contact: <sip:041772XXXX@202.52.129.50> (39)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 5: Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50 (55)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 6: CSeq: 102 INVITE (16)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 7: User-Agent: MaxoTel (19)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 8: Max-Forwards: 70 (16)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 9: Date: Mon, 11 Oct 2010 14:27:46 GMT (35)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 11: Supported: replaces (19)
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: Header 12: Content-Type: application/sdp (29)
[2010-10-12 00:27:46] VERBOSE[23038] logger.c: Reliably Transmitting (NAT) to 27.50.95.170:5060:
INVITE sip:036108YYYY@27.50.95.170:5060 SIP/2.0
Via: SIP/2.0/UDP 202.52.129.50:5060;branch=z9hG4bK7afce819;rport
From: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6
To: <sip:036108YYYY@27.50.95.170:5060>
Contact: <sip:041772XXXX@202.52.129.50>
Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50
CSeq: 102 INVITE
User-Agent: MaxoTel
Max-Forwards: 70
Date: Mon, 11 Oct 2010 14:27:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 21628 21628 IN IP4 202.52.129.50
s=session
c=IN IP4 202.52.129.50
t=0 0
m=audio 18216 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[2010-10-12 00:27:46] DEBUG[23038] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id  #-1
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Driver for channel 'SIP/fromsymbio-00000043' does not support indication 3, emulating it
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Set channel SIP/fromsymbio-00000043 to write format alaw
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Set channel SIP/fromsymbio-00000043 to write format slin
[2010-10-12 00:27:46] DEBUG[23038] channel.c: Prodding channel 'SIP/fromsymbio-00000043'
[2010-10-12 00:27:46] VERBOSE[21634] logger.c: 
<--- SIP read from 27.50.95.170:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 202.52.129.50:5060;branch=z9hG4bK7afce819;rport=5060
From: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6
To: <sip:036108YYYY@27.50.95.170:5060>
Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50
CSeq: 102 INVITE
Server: VoipNow
Content-Length: 0


<------------->
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 0: SIP/2.0 100 Giving a try (24)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 1: Via: SIP/2.0/UDP 202.52.129.50:5060;branch=z9hG4bK7afce819;rport=5060 (69)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 2: From: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6 (64)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 3: To: <sip:036108YYYY@27.50.95.170:5060> (38)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 4: Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50 (55)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 6: Server: VoipNow (15)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 7: Content-Length: 0 (17)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: Header 8:  (0)
[2010-10-12 00:27:46] VERBOSE[21634] logger.c: --- (8 headers 0 lines) ---
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: = Found Their Call ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50 Their Tag  Our tag: as6e5377a6
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: *** SIP TIMER: Cancelling retransmission #40635 - INVITE (got response)
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50' Request 102: Found
[2010-10-12 00:27:46] DEBUG[21634] chan_sip.c: SIP response 100 to standard invite


[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Invalid SIP message - rejected , no callid, len 340
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 0:  (0)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line:  (0)
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: 
<--- SIP read from 27.50.95.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.52.129.50:5060;received=202.52.129.50;branch=z9hG4bK7afce819;rport=5060
Record-Route: <sip:27.50.95.170;lr=on;ftag=as6e5377a6;did=eb6.961b12d5>
From: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6
To: <sip:036108YYYY@27.50.95.170:5060>;tag=as1317806b
Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50
Contact: <sip:036108YYYY@27.50.95.170:5060>
CSeq: 102 INVITE
Server: abc telecommunications
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 335

v=0
o=root 1513662510 1513662510 IN IP4 27.50.95.170
s=Asterisk PBX 1.6.1.20
c=IN IP4 27.50.95.170
t=0 0
m=audio 11178 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 1: Via: SIP/2.0/UDP 202.52.129.50:5060;received=202.52.129.50;branch=z9hG4bK7afce819;rport=5060 (92)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 2: Record-Route: <sip:27.50.95.170;lr=on;ftag=as6e5377a6;did=eb6.961b12d5> (71)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 3: From: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6 (64)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 4: To: <sip:036108YYYY@27.50.95.170:5060>;tag=as1317806b (53)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 5: Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50 (55)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 6: Contact: <sip:036108YYYY@27.50.95.170:5060> (43)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 7: CSeq: 102 INVITE (16)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 8: Server: abc telecommunications (30)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 10: Supported: replaces, timer (26)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 11: Content-Type: application/sdp (29)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 12: Content-Length: 335 (19)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 13:  (0)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: v=0 (3)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: o=root 1513662510 1513662510 IN IP4 27.50.95.170 (48)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: s=Asterisk PBX 1.6.1.20 (23)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: c=IN IP4 27.50.95.170 (21)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: t=0 0 (5)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: m=audio 11178 RTP/AVP 0 8 18 101 (32)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=fmtp:18 annexb=no (19)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=fmtp:101 0-16 (15)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=silenceSupp:off - - - - (25)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=ptime:20 (10)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=sendrecv (10)
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: --- (13 headers 15 lines) ---
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: = Found Their Call ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50 Their Tag  Our tag: as6e5377a6
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Acked pending invite 102
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Stopping retransmission on '0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50' of Request 102: Match Found
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: SIP response 200 to standard invite
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP o=root 1513662510 1513662510 IN IP4 27.50.95.170... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP s=Asterisk PBX 1.6.1.20... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP c=IN IP4 27.50.95.170... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Found RTP audio format 0
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Found RTP audio format 8
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Found RTP audio format 18
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Found RTP audio format 101
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Found audio description format PCMU for ID 0
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Found audio description format PCMA for ID 8
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Found audio description format G729 for ID 18
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Found audio description format telephone-event for ID 101
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: T38 state changed to 0 on channel SIP/74189-00000044
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Our T38 capability = (3872), peer T38 capability (0), joint T38 capability (3872)
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Peer audio RTP is at port 27.50.95.170:11178
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Peer doesn't provide T.38 UDPTL
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: We're settling with these formats: 0x10c (ulaw|alaw|g729)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: We have an owner, now see if we need to change this call
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Updating call counter for outgoing call
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: build_route: Record-Route hop: <sip:27.50.95.170;lr=on;ftag=as6e5377a6;did=eb6.961b12d5>
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: list_route: hop: <sip:27.50.95.170;lr=on;ftag=as6e5377a6;did=eb6.961b12d5>
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: set_destination: Parsing <sip:27.50.95.170;lr=on;ftag=as6e5377a6;did=eb6.961b12d5> for address/port to send to
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: set_destination: set destination to 27.50.95.170, port 5060
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Transmitting (NAT) to 27.50.95.170:5060:
ACK sip:036108YYYY@27.50.95.170:5060 SIP/2.0
Via: SIP/2.0/UDP 202.52.129.50:5060;branch=z9hG4bK11c8c338;rport
Route: <sip:27.50.95.170;lr=on;ftag=as6e5377a6;did=eb6.961b12d5>
From: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6
To: <sip:036108YYYY@27.50.95.170:5060>;tag=as1317806b
Contact: <sip:041772XXXX@202.52.129.50>
Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50
CSeq: 102 ACK
User-Agent: MaxoTel
Max-Forwards: 70
Content-Length: 0


---
[2010-10-12 00:27:48] DEBUG[23038] devicestate.c: Notification of state change to be queued on device/channel SIP/74189
[2010-10-12 00:27:48] DEBUG[23038] pbx.c: Launching 'AGI'
[2010-10-12 00:27:48] DEBUG[21631] devicestate.c: No provider found, checking channel drivers for SIP - 74189
[2010-10-12 00:27:48] DEBUG[21631] chan_sip.c: Checking device state for peer 74189
[2010-10-12 00:27:48] DEBUG[21631] res_config_mysql.c: MySQL RealTime: Everything is fine.
[2010-10-12 00:27:48] DEBUG[21631] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_accounts WHERE name = '74189' AND host = 'dynamic'
[2010-10-12 00:27:48] DEBUG[21631] res_config_mysql.c: MySQL RealTime: Everything is fine.
[2010-10-12 00:27:48] DEBUG[21631] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_accounts WHERE name = '74189'
[2010-10-12 00:27:48] DEBUG[21631] chan_sip.c: Destroying SIP peer 74189
[2010-10-12 00:27:48] DEBUG[21631] devicestate.c: Changing state for SIP/74189 - state 1 (Not in use)
[2010-10-12 00:27:48] DEBUG[21651] app_queue.c: Device 'SIP/74189' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2010-10-12 00:27:48] DEBUG[23038] app_macro.c: Executed application: AGI
[2010-10-12 00:27:48] DEBUG[23038] pbx.c: Launching 'NoOp'
[2010-10-12 00:27:48] DEBUG[23038] app_macro.c: Executed application: NoOp
[2010-10-12 00:27:48] DEBUG[23038] app_dial.c: Macro exited with status 0
[2010-10-12 00:27:48] DEBUG[23038] channel.c: Set channel SIP/fromsymbio-00000043 to write format alaw
[2010-10-12 00:27:48] DEBUG[23038] channel.c: Set channel SIP/fromsymbio-00000043 to read format ulaw
[2010-10-12 00:27:48] DEBUG[23038] channel.c: Set channel SIP/74189-00000044 to read format alaw
[2010-10-12 00:27:48] DEBUG[23038] devicestate.c: Notification of state change to be queued on device/channel SIP/fromsymbio
[2010-10-12 00:27:48] DEBUG[23038] chan_sip.c: SIP answering channel: SIP/fromsymbio-00000043
[2010-10-12 00:27:48] DEBUG[23038] rtp.c: Setting the marker bit due to a source update
[2010-10-12 00:27:48] DEBUG[23038] chan_sip.c: Setting framing from config on incoming call
[2010-10-12 00:27:48] DEBUG[23038] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True
[2010-10-12 00:27:48] DEBUG[23038] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[2010-10-12 00:27:48] DEBUG[23038] chan_sip.c: -- Done with adding codecs to SDP
[2010-10-12 00:27:48] DEBUG[23038] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw)
[2010-10-12 00:27:48] DEBUG[23038] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id  #-1
[2010-10-12 00:27:48] DEBUG[23038] rtp.c: Changing ssrc from 585610135 to 2126196653 due to a source change
[2010-10-12 00:27:48] DEBUG[23038] rtp.c: Changing ssrc from 520517365 to 1501437133 due to a source change
[2010-10-12 00:27:48] DEBUG[23038] rtp.c: Cannot packet2packet bridge - raw formats are incompatible
[2010-10-12 00:27:48] DEBUG[23038] rtp.c: Ooh, format changed from unknown to ulaw
[2010-10-12 00:27:48] DEBUG[23038] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[2010-10-12 00:27:48] DEBUG[21631] devicestate.c: No provider found, checking channel drivers for SIP - fromsymbio
[2010-10-12 00:27:48] DEBUG[21631] chan_sip.c: Checking device state for peer fromsymbio
[2010-10-12 00:27:48] DEBUG[21631] devicestate.c: Changing state for SIP/fromsymbio - state 1 (Not in use)
[2010-10-12 00:27:48] DEBUG[21651] app_queue.c: Device 'SIP/fromsymbio' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: 
<--- SIP read from 27.50.95.170:5060 --->
INVITE sip:041772XXXX@202.52.129.50 SIP/2.0
Via: SIP/2.0/UDP 27.50.95.170:5060;branch=z9hG4bK9c0b.c33df492.0
Max-Forwards: 69
From: <sip:036108YYYY@27.50.95.170:5060>;tag=as1317806b
To: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6
Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50
Contact: <sip:036108YYYY@27.50.95.170:5060>
CSeq: 102 INVITE
User-Agent: abc telecommunications
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 297

v=0
o=root 1513662510 1513662511 IN IP4 27.50.95.170
s=Asterisk PBX 1.6.1.20
c=IN IP4 27.50.95.170
t=0 0
m=image 4371 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 0: INVITE sip:041772XXXX@202.52.129.50 SIP/2.0 (43)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 1: Via: SIP/2.0/UDP 27.50.95.170:5060;branch=z9hG4bK9c0b.c33df492.0 (64)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 2: Max-Forwards: 69 (16)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 3: From: <sip:036108YYYY@27.50.95.170:5060>;tag=as1317806b (55)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 4: To: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6 (62)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 5: Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50 (55)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 6: Contact: <sip:036108YYYY@27.50.95.170:5060> (43)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 7: CSeq: 102 INVITE (16)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 8: User-Agent: abc telecommunications (34)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 10: Supported: replaces, timer (26)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 11: Content-Type: application/sdp (29)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 12: Content-Length: 297 (19)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 13:  (0)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: v=0 (3)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: o=root 1513662510 1513662511 IN IP4 27.50.95.170 (48)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: s=Asterisk PBX 1.6.1.20 (23)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: c=IN IP4 27.50.95.170 (21)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: t=0 0 (5)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: m=image 4371 udptl t38 (22)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=T38FaxVersion:0 (17)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=T38MaxBitRate:14400 (21)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=T38FaxFillBitRemoval (22)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=T38FaxRateManagement:transferredTCF (37)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=T38FaxMaxDatagram:1400 (24)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=T38FaxUdpEC:t38UDPRedundancy (30)
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: --- (13 headers 12 lines) ---
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: = Found Their Call ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50 Their Tag as1317806b Our tag: as6e5377a6
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Begin: parsing SIP "Supported: replaces, timer"
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Found SIP option: -replaces-
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Matched SIP option: replaces
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Found SIP option: -timer-
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Matched SIP option: timer
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Sending to 27.50.95.170 : 5060 (NAT)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP o=root 1513662510 1513662511 IN IP4 27.50.95.170... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP s=Asterisk PBX 1.6.1.20... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP c=IN IP4 27.50.95.170... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Got T.38 offer in SDP in dialog 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: T38 state changed to 3 on channel SIP/74189-00000044
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: FaxVersion: 0
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (image) SDP a=T38FaxVersion:0... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: T38MaxBitRate: 14400
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (image) SDP a=T38MaxBitRate:14400... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: FillBitRemoval
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (image) SDP a=T38FaxFillBitRemoval... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: RateManagement: transferredTCF
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (image) SDP a=T38FaxRateManagement:transferredTCF... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: FaxMaxDatagram: 1400
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (image) SDP a=T38FaxMaxDatagram:1400... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: UDP EC: t38UDPRedundancy
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (image) SDP a=T38FaxUdpEC:t38UDPRedundancy... OK.
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Our T38 capability = (3872), peer T38 capability (16161), joint T38 capability (3873)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Have T.38 but no audio codecs, accepting offer anyway
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Peer T.38 UDPTL is at port 27.50.95.170:4371
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: We're settling with these formats: 0x0 (nothing)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: We have an owner, now see if we need to change this call
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Got a SIP re-invite for call 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: SIP/74189-00000044: This call is UP.... 
[2010-10-12 00:27:48] VERBOSE[21634] logger.c: 
<--- Transmitting (NAT) to 27.50.95.170:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 27.50.95.170:5060;branch=z9hG4bK9c0b.c33df492.0;received=27.50.95.170
From: <sip:036108YYYY@27.50.95.170:5060>;tag=as1317806b
To: "041772XXXX" <sip:041772XXXX@202.52.129.50>;tag=as6e5377a6
Call-ID: 0ef9d79c1912f8693b8a8fb20e3f1781@202.52.129.50
CSeq: 102 INVITE
User-Agent: MaxoTel
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:041772XXXX@202.52.129.50>
Content-Length: 0


<------------>
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Deferring reinvite on SIP '4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3' - It's UDPTL will be redirected to us (IP 202.52.129.50)
[2010-10-12 00:27:48] DEBUG[23038] rtp.c: Setting the marker bit due to a source update



[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Invalid SIP message - rejected , no callid, len 344
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 0: ACK sip:036108YYYY@202.52.129.50 SIP/2.0 (40)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 1: Via: SIP/2.0/UDP 125.213.160.7:5060;branch=z9hG4bK17dd5a0074cb31ee2-cd209-1 (75)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 2: Max-Forwards: 70 (16)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 3: To: <sip:036108YYYY@202.52.129.50:5060>;tag=as678e7007 (54)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 4: From: "041772XXXX"<sip:041772XXXX@125.213.160.7:5060>;tag=5c65bd79-co8402-INS001 (80)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 5: Call-ID: 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3 (58)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 6: CSeq: 840201 ACK (16)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 7: User-Agent: ENSR2.5.4 (21)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 8: Content-Length: 0 (17)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 9:  (0)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: = Found Their Call ID: 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3 Their Tag 5c65bd79-co8402-INS001 Our tag: as678e7007
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #40679
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Stopping retransmission on '4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3' of Response 840201: Match Found
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Sending pending reinvite on '4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3'
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Strict routing enforced for session 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: -- Done with adding codecs to SDP
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Initializing already initialized SIP dialog 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3 (presumably reinvite)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 0: INVITE sip:041772XXXX@125.213.160.7:5060 SIP/2.0 (48)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 1: Via: SIP/2.0/UDP 202.52.129.50:5060;branch=z9hG4bK7c18d2bd;rport (64)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 2: From: <sip:036108YYYY@202.52.129.50:5060>;tag=as678e7007 (56)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 3: To: "041772XXXX"<sip:041772XXXX@125.213.160.7:5060>;tag=5c65bd79-co8402-INS001 (78)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 4: Contact: <sip:036108YYYY@202.52.129.50> (39)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 5: Call-ID: 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3 (58)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 6: CSeq: 102 INVITE (16)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 7: User-Agent: MaxoTel (19)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 8: Max-Forwards: 70 (16)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 10: Supported: replaces (19)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 12: Content-Type: application/sdp (29)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id  #-1

[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Invalid SIP message - rejected , no callid, len 338
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 1: Via: SIP/2.0/UDP 202.52.129.50:5060;branch=z9hG4bK7c18d2bd;rport (64)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 2: Contact: <sip:041772XXXX@125.213.160.7:5060> (44)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 3: To: <sip:041772XXXX@125.213.160.7:5060>;tag=5c65bd79-co8402-INS001 (66)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 4: From: <sip:036108YYYY@202.52.129.50:5060>;tag=as678e7007 (56)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 5: Call-ID: 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3 (58)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 6: CSeq: 102 INVITE (16)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 7: Content-Type: application/sdp (29)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 8: User-Agent: ENSR2.5.4 (21)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 9: Content-Length: 239 (19)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Header 10:  (0)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: v=0 (3)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: o=- 1550171513 1550171514 IN IP4 125.213.160.7 (46)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: s=ENSResip (10)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: c=IN IP4 125.213.160.11 (23)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: t=0 0 (5)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: m=audio 17090 RTP/AVP 8 101 (27)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=fmtp:101 0-15 (15)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=silenceSupp:off - - - - (25)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Line: a=sendrecv (10)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: = Found Their Call ID: 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3 Their Tag 5c65bd79-co8402-INS001 Our tag: as678e7007
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Acked pending invite 102
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #40682
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Stopping retransmission on '4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3' of Request 102: Match Found
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: SIP response 200 to RE-invite on outgoing call 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP o=- 1550171513 1550171514 IN IP4 125.213.160.7... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP s=ENSResip... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP c=IN IP4 125.213.160.11... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: T38 state changed to 0 on channel SIP/fromsymbio-00000043
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: We're settling with these formats: 0x8 (alaw)
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: We have an owner, now see if we need to change this call
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Updating call counter for incoming call
[2010-10-12 00:27:48] DEBUG[21634] chan_sip.c: Strict routing enforced for session 4a98-4b7-9112010142746-img-01-mas-0-125.213.168.3

(CALLER HEARS SILENCE, HANGS UP).