<--- SIP read from 212.147.20.222:5060 --->
INVITE sip:00090041625037902@194.148.41.50 SIP/2.0
Via: SIP/2.0/UDP 212.147.20.222:5060;branch=z9hG4bK1c77a5a54
Max-Forwards: 70
Content-Length: 334
To: sip:00090041625037902@194.148.41.50
From: sip:00090041217211111@212.147.20.222:5060;tag=c2a0dce8c1c760b
Call-ID: 2326fdfcf172a00f9152e3d75036f6b0@212.147.20.222
CSeq: 423024424 INVITE
Route: <sip:194.148.41.50;lr>
Supported: timer
Remote-Party-ID: <sip:00090041217211111@212.147.20.222:5060>;party=calling;screen=yes;privacy=off
Content-Type: application/sdp
Contact: sip:00090041217211111@212.147.20.222:5060
Supported: replaces
User-Agent: Patton SN2400 MxSF v3.2.8.45 00A0BA01C558

v=0
o=MxSIP 0 26389 IN IP4 212.147.20.222
s=SIP Call
c=IN IP4 212.147.20.222
t=0 0
m=audio 5302 RTP/AVP 18 8 0 100
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:18 annexb=no
a=fmtp:100 192-194
a=sendrecv
m=image 5304 udptl t38
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv

<------------->

[Mar 28 16:27:19] --- (15 headers 16 lines) ---
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:2582 do_setnat: Setting NAT on RTP to Off
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:2592 do_setnat: Setting NAT on UDPTL to Off
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:4317 sip_alloc: Allocating new SIP dialog for 2326fdfcf172a00f9152e3d75036f6b0@212.147.20.222 - INVITE (With RTP)
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:14640 handle_request: **** Received INVITE (5) - Command in SIP INVITE
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:1680 parse_sip_options: Begin: parsing SIP "Supported: timer"
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:1688 parse_sip_options: Found SIP option: -timer-
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:1694 parse_sip_options: Matched SIP option: timer
[Mar 28 16:27:19] Sending to 212.147.20.222 : 5060 (no NAT)
[Mar 28 16:27:19] Using INVITE request as basis request - 2326fdfcf172a00f9152e3d75036f6b0@212.147.20.222
[Mar 28 16:27:19] DEBUG[18254]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine.
[Mar 28 16:27:19] DEBUG[18254]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '00090041217211111'
[Mar 28 16:27:19] Found peer 'sip_trunk_gva-mg-01'
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:2582 do_setnat: Setting NAT on RTP to Off
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:2592 do_setnat: Setting NAT on UDPTL to Off
[Mar 28 16:27:19] Found RTP audio format 18
[Mar 28 16:27:19] Found RTP audio format 8
[Mar 28 16:27:19] Found RTP audio format 0
[Mar 28 16:27:19] Found RTP audio format 100
[Mar 28 16:27:19] Got T.38 offer in SDP in dialog 2326fdfcf172a00f9152e3d75036f6b0@212.147.20.222
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:4852 process_sdp: T38 state changed to 3 on channel <none>
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:4902 process_sdp: Peer T.38 UDPTL is at port 212.147.20.222:5304
[Mar 28 16:27:19] Peer audio RTP is at port 212.147.20.222:5302
[Mar 28 16:27:19] Found description format G729 for ID 18
[Mar 28 16:27:19] Found description format PCMA for ID 8
[Mar 28 16:27:19] Found description format PCMU for ID 0
[Mar 28 16:27:19] Found description format X-NSE for ID 100
[Mar 28 16:27:19] Got unsupported a:fmtp in SDP offer 
[Mar 28 16:27:19] Got unsupported a:fmtp in SDP offer 
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:5109 process_sdp: UDP EC: t38UDPRedundancy
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:5129 process_sdp: Our T38 capability = (3856), peer T38 capability (32), joint T38 capability (32)
[Mar 28 16:27:19] Capabilities: us - 0x108 (alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
[Mar 28 16:27:19] Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Mar 28 16:27:19] Peer audio RTP is at port 212.147.20.222:5302
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:5216 process_sdp: We're settling with these formats: 0x108 (alaw|g729)
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:13408 handle_request_invite: Checking SIP call limits for device 
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:3010 update_call_counter: Updating call counter for incoming call
[Mar 28 16:27:19] Looking for 00090041625037902 in default (domain 194.148.41.50)
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:3812 sip_new: *** Our native formats are 0x100 (g729) 
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:3813 sip_new: *** Joint capabilities are 0x108 (alaw|g729) 
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:3814 sip_new: *** Our capabilities are 0x108 (alaw|g729) 
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:3815 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729) 
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:3838 sip_new: This channel will not be able to handle video.
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:7987 build_route: build_route: Contact hop: sip:00090041217211111@212.147.20.222:5060
[Mar 28 16:27:19] list_route: hop: <sip:00090041217211111@212.147.20.222:5060>
[Mar 28 16:27:19] DEBUG[18254]: chan_sip.c:13483 handle_request_invite: SIP/5060-006ff080: New call is still down.... Trying... 
[Mar 28 16:27:19] 




[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:2852 sip_call: Our T38 capability (3856), joint T38 capability (3856)
[Mar 28 16:27:19] DEBUG[18231]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 0625037902
[Mar 28 16:27:19] DEBUG[18231]: chan_sip.c:15251 sip_devicestate: Checking device state for peer 0625037902
[Mar 28 16:27:19] DEBUG[18231]: devicestate.c:287 do_state_change: Changing state for SIP/0625037902 - state 6 (Ringing)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:6195 add_sdp: ** Our capability: 0x10e (gsm|ulaw|alaw|g729) Video flag: False
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:6196 add_sdp: ** Our prefcodec: 0x100 (g729) 
[Mar 28 16:27:19] DEBUG[18262]: app_queue.c:546 changethread: Device 'SIP/0625037902' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Mar 28 16:27:19] Audio is at 194.148.41.50 port 18714
[Mar 28 16:27:19] Adding codec 0x100 (g729) to SDP
[Mar 28 16:27:19] Adding codec 0x2 (gsm) to SDP
[Mar 28 16:27:19] Adding codec 0x8 (alaw) to SDP
[Mar 28 16:27:19] Adding codec 0x4 (ulaw) to SDP
[Mar 28 16:27:19] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:6327 add_sdp: -- Done with adding codecs to SDP
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:6372 add_sdp: Done building SDP. Settling with this capability: 0x10e (gsm|ulaw|alaw|g729)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 0: INVITE sip:0625037902@10.0.0.102;user=phone SIP/2.0 (51)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 1: Via: SIP/2.0/UDP 194.148.41.50:5060;branch=z9hG4bK02e1701d;rport (64)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 2: From: "0217211111" <sip:0217211111@194.148.41.50>;tag=as06ee4aee (64)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 3: To: <sip:0625037902@10.0.0.102;user=phone> (42)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 4: Contact: <sip:0217211111@194.148.41.50> (39)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 5: Call-ID: 27cc07382a1339b57fb1d47c367e4114@194.148.41.50 (55)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 6: CSeq: 102 INVITE (16)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 7: User-Agent: VTX SIP (19)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 8: Max-Forwards: 70 (16)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 9: Date: Wed, 28 Mar 2007 14:27:19 GMT (35)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 11: Supported: replaces (19)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 12: Content-Type: application/sdp (29)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 13: Content-Length: 336 (19)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4580 parse_request: Header 14:  (0)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: v=0 (3)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: o=root 18226 18226 IN IP4 194.148.41.50 (39)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: s=session (9)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: c=IN IP4 194.148.41.50 (22)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: t=0 0 (5)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: m=audio 18714 RTP/AVP 18 3 8 0 101 (34)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=fmtp:18 annexb=no (19)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=rtpmap:3 GSM/8000 (19)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=fmtp:101 0-16 (15)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=silenceSupp:off - - - - (25)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=ptime:20 (10)
[Mar 28 16:27:19] DEBUG[18260]: chan_sip.c:4612 parse_request: Line: a=sendrecv (10)
[Mar 28 16:27:19] Reliably Transmitting (NAT) to 212.147.65.170:8157:
INVITE sip:0625037902@10.0.0.102;user=phone SIP/2.0
Via: SIP/2.0/UDP 194.148.41.50:5060;branch=z9hG4bK02e1701d;rport
From: "0217211111" <sip:0217211111@194.148.41.50>;tag=as06ee4aee
To: <sip:0625037902@10.0.0.102;user=phone>
Contact: <sip:0217211111@194.148.41.50>
Call-ID: 27cc07382a1339b57fb1d47c367e4114@194.148.41.50
CSeq: 102 INVITE
User-Agent: VTX SIP
Max-Forwards: 70
Date: Wed, 28 Mar 2007 14:27:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 18226 18226 IN IP4 194.148.41.50
s=session
c=IN IP4 194.148.41.50
t=0 0
m=audio 18714 RTP/AVP 18 3 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<--- SIP read from 212.147.65.170:8157 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.148.41.50:5060;branch=z9hG4bK02e1701d;rport
From: "0217211111" <sip:0217211111@194.148.41.50>;tag=as06ee4aee
To: <sip:0625037902@10.0.0.102;user=phone>;tag=bc2dffff76240000
Call-ID: 27cc07382a1339b57fb1d47c367e4114@194.148.41.50
CSeq: 102 INVITE
User-Agent: Grandstream HT488 1.0.3.44
Contact: <sip:0625037902@10.0.0.102;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 159

v=0
o=0625037902 8000 8000 IN IP4 10.0.0.102
s=SIP Call
c=IN IP4 10.0.0.102
t=0 0
m=audio 5004 RTP/AVP 18
a=sendrecv
a=rtpmap:18 G729/8000
a=ptime:20

<------------->

[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4580 parse_request: Header 11: Content-Length: 159 (19)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4580 parse_request: Header 12:  (0)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4612 parse_request: Line: v=0 (3)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4612 parse_request: Line: o=0625037902 8000 8000 IN IP4 10.0.0.102 (40)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4612 parse_request: Line: s=SIP Call (10)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4612 parse_request: Line: c=IN IP4 10.0.0.102 (19)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4612 parse_request: Line: t=0 0 (5)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4612 parse_request: Line: m=audio 5004 RTP/AVP 18 (23)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4612 parse_request: Line: a=sendrecv (10)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4612 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4612 parse_request: Line: a=ptime:20 (10)
[Mar 28 16:27:21] --- (12 headers 9 lines) ---
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:2071 __sip_ack: Acked pending invite 102
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:2089 __sip_ack: Stopping retransmission on '27cc07382a1339b57fb1d47c367e4114@194.148.41.50' of Request 102: Match Not Found
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:11648 handle_response_invite: SIP response 200 to standard invite
[Mar 28 16:27:21] Found RTP audio format 18
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:4906 process_sdp: Peer doesn't provide T.38 UDPTL
[Mar 28 16:27:21] Peer audio RTP is at port 10.0.0.102:5004
[Mar 28 16:27:21] Found description format G729 for ID 18
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:5136 process_sdp: T38 state changed to 0 on channel SIP/0625037902-00716940
[Mar 28 16:27:21] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
[Mar 28 16:27:21] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Mar 28 16:27:21] Peer audio RTP is at port 10.0.0.102:5004
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:5216 process_sdp: We're settling with these formats: 0x100 (g729)
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:5223 process_sdp: We have an owner, now see if we need to change this call
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:3010 update_call_counter: Updating call counter for outgoing call
[Mar 28 16:27:21] DEBUG[18254]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/0625037902
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:7987 build_route: build_route: Contact hop: <sip:0625037902@10.0.0.102;user=phone>
[Mar 28 16:27:21] list_route: hop: <sip:0625037902@10.0.0.102;user=phone>
[Mar 28 16:27:21] DEBUG[18231]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 0625037902
[Mar 28 16:27:21] DEBUG[18231]: chan_sip.c:15251 sip_devicestate: Checking device state for peer 0625037902
[Mar 28 16:27:21] DEBUG[18254]: chan_sip.c:5650 reqprep: Strict routing enforced for session 27cc07382a1339b57fb1d47c367e4114@194.148.41.50
[Mar 28 16:27:21] DEBUG[18231]: devicestate.c:287 do_state_change: Changing state for SIP/0625037902 - state 2 (In use)
[Mar 28 16:27:21] set_destination: Parsing <sip:0625037902@10.0.0.102;user=phone> for address/port to send to
[Mar 28 16:27:21] set_destination: set destination to 10.0.0.102, port 5060
[Mar 28 16:27:21] DEBUG[18264]: app_queue.c:546 changethread: Device 'SIP/0625037902' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Mar 28 16:27:21] Transmitting (NAT) to 212.147.65.170:8157:

[Mar 28 16:27:21]     -- Call on SIP/0625037902-00716940 left from hold
[Mar 28 16:27:21] DEBUG[18260]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/0625037902-00716940
[Mar 28 16:27:21]     -- SIP/0625037902-00716940 answered SIP/5060-006ff080
[Mar 28 16:27:21] DEBUG[18231]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 0625037902
[Mar 28 16:27:21] DEBUG[18231]: chan_sip.c:15251 sip_devicestate: Checking device state for peer 0625037902
[Mar 28 16:27:21] DEBUG[18231]: devicestate.c:287 do_state_change: Changing state for SIP/0625037902 - state 2 (In use)
[Mar 28 16:27:21] DEBUG[18260]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/5060-006ff080
[Mar 28 16:27:21] DEBUG[18265]: app_queue.c:546 changethread: Device 'SIP/0625037902' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Mar 28 16:27:21] DEBUG[18260]: chan_sip.c:3470 sip_answer: SIP answering channel: SIP/5060-006ff080
[Mar 28 16:27:21] DEBUG[18260]: chan_sip.c:3474 sip_answer: T38State change to 5 on channel SIP/5060-006ff080
[Mar 28 16:27:21] DEBUG[18231]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 5060
[Mar 28 16:27:21] DEBUG[18231]: chan_sip.c:15251 sip_devicestate: Checking device state for peer 5060
[Mar 28 16:27:21] DEBUG[18260]: chan_sip.c:6049 add_t38_sdp: T.38 UDPTL is at 194.148.41.50 port 15640
[Mar 28 16:27:21] DEBUG[18260]: chan_sip.c:6055 add_t38_sdp: Our T38 capability (3856), peer T38 capability (32), joint capability (32)
[Mar 28 16:27:21] DEBUG[18260]: chan_sip.c:5998 t38_get_rate: Strange, T38MaxFaxRate NOT found in peers T38 SDP.
[Mar 28 16:27:21] 
<--- Reliably Transmitting (no NAT) to 212.147.20.222:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.147.20.222:5060;branch=z9hG4bK1c77a5a54;received=212.147.20.222
From: sip:00090041217211111@212.147.20.222:5060;tag=c2a0dce8c1c760b
To: sip:00090041625037902@194.148.41.50;tag=as1fe29760
Call-ID: 2326fdfcf172a00f9152e3d75036f6b0@212.147.20.222
CSeq: 423024424 INVITE
User-Agent: VTX SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:00090041625037902@194.148.41.50>
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 18226 18226 IN IP4 194.148.41.50
s=session
c=IN IP4 194.148.41.50
t=0 0
m=image 15640 udptl t38
a=T38FaxVersion:0
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

<------------>
[Mar 28 16:27:21] DEBUG[18260]: chan_sip.c:1975 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #35
[Mar 28 16:27:21] DEBUG[18231]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine.
[Mar 28 16:27:21] DEBUG[18231]: res_config_mysql.c:139 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '5060'
[Mar 28 16:27:21] DEBUG[18231]: devicestate.c:287 do_state_change: Changing state for SIP/5060 - state 4 (Invalid)
[Mar 28 16:27:21] DEBUG[18266]: app_queue.c:546 changethread: Device 'SIP/5060' changed to state '4' (Invalid) but we don't care because they're not a member of any queue.
[Mar 28 16:27:21] DEBUG[18260]: rtp.c:1151 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 212.147.65.170:12697
[Mar 28 16:27:21] DEBUG[18260]: rtp.c:2689 ast_rtp_write: Ooh, format changed from unknown to g729
[Mar 28 16:27:21] DEBUG[18260]: rtp.c:2706 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20
[Mar 28 16:27:21] 
<--- SIP read from 212.40.9.10:5060 --->
OPTIONS sip:212.40.9.9:5060 SIP/2.0
Via: SIP/2.0/UDP bigip.vtx.ch:5060;branch=z9hG4bK35763c11
Max-Forwards: 1
To: sip:212.40.9.9
From: sip:anonymous@bigip.vtx.ch;tag=6b598fdb
Call-ID: 0000216b460a77af@bigip.vtx.ch
CSeq: 1175091119 OPTIONS
Accept: application/sdp
Content-Length: 0





<------------>
[Mar 28 16:27:21] Scheduling destruction of SIP dialog '0000216b460a77af@bigip.vtx.ch' in 32000 ms (Method: OPTIONS)
[Mar 28 16:27:22] DEBUG[18254]: chan_sip.c:1866 retrans_pkt: SIP TIMER: Rescheduling retransmission #35 (1) SIP/2.0 - 1
[Mar 28 16:27:22] DEBUG[18254]: chan_sip.c:1880 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #35)) 
[Mar 28 16:27:22] Retransmitting #1 (no NAT) to 212.147.20.222:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.147.20.222:5060;branch=z9hG4bK1c77a5a54;received=212.147.20.222
From: sip:00090041217211111@212.147.20.222:5060;tag=c2a0dce8c1c760b
To: sip:00090041625037902@194.148.41.50;tag=as1fe29760
Call-ID: 2326fdfcf172a00f9152e3d75036f6b0@212.147.20.222
CSeq: 423024424 INVITE
User-Agent: VTX SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:00090041625037902@194.148.41.50>
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 18226 18226 IN IP4 194.148.41.50
s=session
c=IN IP4 194.148.41.50
t=0 0
m=image 15640 udptl t38
a=T38FaxVersion:0
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

---
[Mar 28 16:27:22] 
<--- SIP read from 212.147.20.222:5060 --->
ACK sip:00090041625037902@194.148.41.50 SIP/2.0
Via: SIP/2.0/UDP 212.147.20.222:5060;branch=z9hG4bKfb0aea80b
Max-Forwards: 70
Content-Length: 0
To: sip:00090041625037902@194.148.41.50;tag=as1fe29760
From: sip:00090041217211111@212.147.20.222:5060;tag=c2a0dce8c1c760b
Call-ID: 2326fdfcf172a00f9152e3d75036f6b0@212.147.20.222
CSeq: 423024424 ACK
Contact: sip:00090041217211111@212.147.20.222:5060
User-Agent: Patton SN2400 MxSF v3.2.8.45 00A0BA01C558