

<--- SIP read from UDP:192.168.165.33:5060 --->
REGISTER sip:192.168.119.227 SIP/2.0
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac972240673
Max-Forwards: 70
From: <sip:221119966@192.168.119.227>;tag=1c972234823
To: <sip:221119966@192.168.119.227>
Call-ID: 207566505121200020742@192.168.165.33
CSeq: 119 REGISTER
Contact: <sip:221119966@192.168.165.33:5060>;expires=600
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 600
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
[Jan 23 13:29:33] DEBUG[3378]: acl.c:507 ast_ouraddrfor: Found IP address for this socket
[Jan 23 13:29:33] DEBUG[3378]: chan_sip.c:3725 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.119.227:5060
[Jan 23 13:29:33] DEBUG[3378]: chan_sip.c:7400 sip_alloc: Allocating new SIP dialog for 207566505121200020742@192.168.165.33 - REGISTER (No RTP)
Sending to 192.168.165.33 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.165.33:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac972240673;received=192.168.165.33
From: <sip:221119966@192.168.119.227>;tag=1c972234823
To: <sip:221119966@192.168.119.227>;tag=as03568e6c
Call-ID: 207566505121200020742@192.168.165.33
CSeq: 119 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a1aaa5a"
Content-Length: 0


<------------>
[Jan 23 13:29:33] DEBUG[3378]: chan_sip.c:3604 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.165.33:5060
Scheduling destruction of SIP dialog '207566505121200020742@192.168.165.33' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.165.33:5060 --->
REGISTER sip:192.168.119.227 SIP/2.0
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac972500931
Max-Forwards: 70
From: <sip:221119966@192.168.119.227>;tag=1c972234823
To: <sip:221119966@192.168.119.227>
Call-ID: 207566505121200020742@192.168.165.33
CSeq: 120 REGISTER
Authorization: Digest username="221119966",realm="asterisk",nonce="7a1aaa5a",uri="sip:192.168.119.227",algorithm=MD5,response="d15cdb8b0c91503318afe88a89587870"
Contact: <sip:221119966@192.168.165.33:5060>;expires=600
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 600
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.165.33 : 5060 (no NAT)
[Jan 23 13:29:33] DEBUG[3378]: chan_sip.c:12531 parse_register_contact: Store REGISTER's Contact header for call routing.

<--- Transmitting (no NAT) to 192.168.165.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac972500931;received=192.168.165.33
From: <sip:221119966@192.168.119.227>;tag=1c972234823
To: <sip:221119966@192.168.119.227>;tag=as03568e6c
Call-ID: 207566505121200020742@192.168.165.33
CSeq: 120 REGISTER
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 600
Contact: <sip:221119966@192.168.165.33:5060>;expires=600
Date: Sun, 23 Jan 2011 12:29:33 GMT
Content-Length: 0


<------------>
[Jan 23 13:29:33] DEBUG[3378]: chan_sip.c:3604 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.165.33:5060
Scheduling destruction of SIP dialog '207566505121200020742@192.168.165.33' in 32000 ms (Method: REGISTER)
[Jan 23 13:29:33] DEBUG[3350]: chan_sip.c:23302 sip_devicestate: Checking device state for peer 221119966
[Jan 23 13:29:33] DEBUG[3350]: devicestate.c:462 do_state_change: Changing state for SIP/221119966 - state 1 (Not in use)
[Jan 23 13:29:33] DEBUG[3350]: devicestate.c:442 devstate_event: device 'SIP/221119966' state '1'
[Jan 23 13:29:33] DEBUG[3379]: app_queue.c:1092 handle_statechange: Device 'SIP/221119966' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
obelix*CLI>
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:2943 _sip_tcp_helper_thread: Starting thread for TCP server

<--- SIP read from TCP:192.168.119.240:65306 --->
INVITE sip:221119966@192.168.119.227;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.119.240;branch=z9hG4bKac1595438338;alias
Max-Forwards: 70
From: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
To: <sip:221119966@192.168.119.227;user=phone>
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 1 INVITE
Contact: <sip:221119923@192.168.119.240>
Supported: em,100rel,timer,replaces,path,histinfo,early-session,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Remote-Party-ID: "Wywolanie" <sip:221119923@192.168.119.227>;party=calling;privacy=off;screen=yes;screen-ind=3;npi=1;ton=2
Remote-Party-ID: <sip:221119966@192.168.119.227>;party=called;npi=0;ton=0
x-channel: ds/ds1-1/2;IP=192.168.119.240
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.012.005
Privacy: none
P-Asserted-Identity: "Wywolanie" <sip:221119923@audiocodes.com>
History-Info: <sip:221119966@192.168.119.227;user=phone>;index=1
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 287

v=0
o=AudiocodesGW 1595405269 1595404952 IN IP4 192.168.119.240
s=Phone-Call
c=IN IP4 192.168.119.240
t=0 0
m=audio 10970 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (20 headers 13 lines) ---
[Jan 23 13:30:01] DEBUG[3413]: acl.c:507 ast_ouraddrfor: Found IP address for this socket
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3725 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_TCP with address 192.168.119.227:5060
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:5073 do_setnat: Setting NAT on RTP to Off
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:5081 do_setnat: Setting NAT on UDPTL to Off
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:7400 sip_alloc: Allocating new SIP dialog for 1595423211231201113301@192.168.119.240 - INVITE (With RTP)
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3418 parse_sip_options: Begin: parsing SIP "Supported: em,100rel,timer,replaces,path,histinfo,early-session,resource-priority,sdp-anat"
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3426 parse_sip_options: Found SIP option: -em-
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3448 parse_sip_options: Found no match for SIP option: em (Please file bug report!)
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3426 parse_sip_options: Found SIP option: -100rel-
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3432 parse_sip_options: Matched SIP option: 100rel
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3426 parse_sip_options: Found SIP option: -timer-
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3432 parse_sip_options: Matched SIP option: timer
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3426 parse_sip_options: Found SIP option: -replaces-
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3432 parse_sip_options: Matched SIP option: replaces
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3426 parse_sip_options: Found SIP option: -path-
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3432 parse_sip_options: Matched SIP option: path
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3426 parse_sip_options: Found SIP option: -histinfo-
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3432 parse_sip_options: Matched SIP option: histinfo
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3426 parse_sip_options: Found SIP option: -early-session-
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3432 parse_sip_options: Matched SIP option: early-session
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3426 parse_sip_options: Found SIP option: -resource-priority-
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3432 parse_sip_options: Matched SIP option: resource-priority
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3426 parse_sip_options: Found SIP option: -sdp-anat-
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3432 parse_sip_options: Matched SIP option: sdp-anat
Sending to 192.168.119.240 : 5060 (no NAT)
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:20183 handle_request_invite: Initializing initreq for method INVITE - callid 1595423211231201113301@192.168.119.240
Using INVITE request as basis request - 1595423211231201113301@192.168.119.240
Found peer 'gateway-vg2' for '221119923' from 192.168.119.240:65306
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:5073 do_setnat: Setting NAT on RTP to Off
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:5081 do_setnat: Setting NAT on UDPTL to Off
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP o=AudiocodesGW 1595405269 1595404952 IN IP4 192.168.119.240... UNSUPPORTED.
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP s=Phone-Call... UNSUPPORTED.
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP c=IN IP4 192.168.119.240... OK.
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Found audio description format G729 for ID 18
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
Found audio description format PCMA for ID 8
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
Found audio description format telephone-event for ID 101
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK.
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.119.240:10970
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8696 process_sdp: Peer doesn't provide T.38 UDPTL
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:8706 process_sdp: We're settling with these formats: 0x108 (alaw|g729)
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:20285 handle_request_invite: Checking SIP call limits for device
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:5715 update_call_counter: Updating call counter for incoming call
Looking for 221119966 in remote (domain 192.168.119.227)
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:6756 sip_new: *** Our native formats are 0x100 (g729)
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:6757 sip_new: *** Joint capabilities are 0x108 (alaw|g729)
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:6758 sip_new: *** Our capabilities are 0x108 (alaw|g729)
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:6759 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729)
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:6789 sip_new: This channel will not be able to handle video.
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:12715 build_route: build_route: Contact hop: <sip:221119923@192.168.119.240>
list_route: hop: <sip:221119923@192.168.119.240>
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:20544 handle_request_invite: SIP/gateway-vg2-00000000: New call is still down.... Trying...

<--- Transmitting (no NAT) to 192.168.119.240:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.119.240;branch=z9hG4bKac1595438338;alias;received=192.168.119.240
From: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
To: <sip:221119966@192.168.119.227;user=phone>
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:221119966@192.168.119.227;transport=TCP>
Content-Length: 0


<------------>
[Jan 23 13:30:01] DEBUG[3413]: chan_sip.c:3604 __sip_xmit: Trying to put 'SIP/2.0 100' onto TCP socket destined for 192.168.119.240:5060
[Jan 23 13:30:01] DEBUG[3350]: chan_sip.c:23302 sip_devicestate: Checking device state for peer gateway-vg2
[Jan 23 13:30:01] DEBUG[3350]: devicestate.c:462 do_state_change: Changing state for SIP/gateway-vg2 - state 1 (Not in use)
[Jan 23 13:30:01] DEBUG[3350]: devicestate.c:442 devstate_event: device 'SIP/gateway-vg2' state '1'
[Jan 23 13:30:01] DEBUG[3379]: app_queue.c:1092 handle_statechange: Device 'SIP/gateway-vg2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Jan 23 13:30:01] DEBUG[3414]: pbx.c:3708 pbx_extension_helper: Launching 'Set'
    -- Executing [221119966@remote:1] Set("SIP/gateway-vg2-00000000", "__TRANSFER_CONTEXT=xxx_test") in new stack
[Jan 23 13:30:01] DEBUG[3414]: pbx.c:3708 pbx_extension_helper: Launching 'Goto'
    -- Executing [221119966@remote:2] Goto("SIP/gateway-vg2-00000000", "xxx_test_in,221119966,1") in new stack
    -- Goto (xxx_test_in,221119966,1)
[Jan 23 13:30:01] DEBUG[3414]: pbx.c:3708 pbx_extension_helper: Launching 'Dial'
    -- Executing [221119966@xxx_test_in:1] Dial("SIP/gateway-vg2-00000000", "SIP/221119966,50") in new stack
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:23396 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729)
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:7400 sip_alloc: Allocating new SIP dialog for 4e42149b21a00ea3530ff3d505a9ffb8@10.100.13.66 - INVITE (With RTP)
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:5073 do_setnat: Setting NAT on RTP to Off
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:5081 do_setnat: Setting NAT on UDPTL to Off
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:3372 obproxy_get: OBPROXY: Not applying OBproxy to this call
[Jan 23 13:30:01] DEBUG[3414]: acl.c:507 ast_ouraddrfor: Found IP address for this socket
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:3725 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.119.227:5060
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:6756 sip_new: *** Our native formats are 0x100 (g729)
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:6757 sip_new: *** Joint capabilities are 0x100 (g729)
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:6758 sip_new: *** Our capabilities are 0x108 (alaw|g729)
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:6759 sip_new: *** AST_CODEC_CHOOSE formats are 0x100 (g729)
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:6761 sip_new: *** Our preferred formats from the incoming channel are 0x100 (g729)
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:6789 sip_new: This channel will not be able to handle video.
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:25701 sip_set_rtp_peer: Early remote bridge setting SIP '5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227' - Sending media to 192.168.119.240
[Jan 23 13:30:01] DEBUG[3414]: rtp.c:2269 ast_rtp_make_compatible: Seeded SDP of 'SIP/221119966-00000001' with that of 'SIP/gateway-vg2-00000000'
[Jan 23 13:30:01] DEBUG[3414]: channel.c:4648 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Jan 23 13:30:01] DEBUG[3414]: channel.c:4648 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Jan 23 13:30:01] DEBUG[3414]: channel.c:4648 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Jan 23 13:30:01] DEBUG[3414]: channel.c:4648 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Jan 23 13:30:01] DEBUG[3414]: channel.c:4648 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Jan 23 13:30:01] DEBUG[3414]: channel.c:4644 ast_channel_inherit_variables: Copying hard-transferable variable TRANSFER_CONTEXT.
[Jan 23 13:30:01] DEBUG[3414]: channel.c:4648 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Jan 23 13:30:01] DEBUG[3414]: channel.c:4648 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Jan 23 13:30:01] DEBUG[3414]: channel.c:4648 ast_channel_inherit_variables: Not copying variable SIPURI.
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:5500 sip_call: Outgoing Call for 221119966
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:5715 update_call_counter: Updating call counter for outgoing call
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:10291 add_sdp: ** Our capability: 0x100 (g729) Video flag: False Text flag: False
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:10292 add_sdp: ** Our prefcodec: 0x100 (g729)
Audio is at 192.168.119.227 port 14782
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:10403 add_sdp: -- Done with adding codecs to SDP
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:10527 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729)
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:3290 initialize_initreq: Initializing initreq for method INVITE - callid 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
Reliably Transmitting (no NAT) to 192.168.165.33:5060:
INVITE sip:221119966@192.168.165.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.119.227:5060;branch=z9hG4bK61f53d2d;rport
Max-Forwards: 70
From: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
To: <sip:221119966@192.168.165.33:5060>
Contact: <sip:221119923@192.168.119.227>
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.16.1
Date: Sun, 23 Jan 2011 12:30:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 295

v=0
o=root 1581035008 1581035008 IN IP4 192.168.119.240
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 192.168.119.240
t=0 0
m=audio 10970 RTP/AVP 18 101
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

---
[Jan 23 13:30:01] DEBUG[3414]: chan_sip.c:3604 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.165.33:5060
    -- Called 221119966

<--- SIP read from UDP:192.168.165.33:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.119.227:5060;branch=z9hG4bK61f53d2d;rport
From: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
To: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 102 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
[Jan 23 13:30:02] DEBUG[3378]: chan_sip.c:4194 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227' Request 102: Found

<--- SIP read from UDP:192.168.165.33:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.119.227:5060;branch=z9hG4bK61f53d2d;rport
From: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
To: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 102 INVITE
Contact: <sip:221119966@192.168.165.33:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
[Jan 23 13:30:02] DEBUG[3378]: chan_sip.c:4194 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227' Request 102: Found
[Jan 23 13:30:02] DEBUG[3350]: chan_sip.c:23302 sip_devicestate: Checking device state for peer 221119966
[Jan 23 13:30:02] DEBUG[3350]: devicestate.c:462 do_state_change: Changing state for SIP/221119966 - state 1 (Not in use)
[Jan 23 13:30:02] DEBUG[3350]: devicestate.c:442 devstate_event: device 'SIP/221119966' state '1'
    -- SIP/221119966-00000001 is ringing
[Jan 23 13:30:02] DEBUG[3414]: chan_sip.c:25701 sip_set_rtp_peer: Early remote bridge setting SIP '1595423211231201113301@192.168.119.240' - Sending media to 192.168.119.227
[Jan 23 13:30:02] DEBUG[3414]: rtp.c:2196 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/gateway-vg2-00000000' with that of 'SIP/221119966-00000001'
[Jan 23 13:30:02] DEBUG[3379]: app_queue.c:1092 handle_statechange: Device 'SIP/221119966' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

<--- Transmitting (no NAT) to 192.168.119.240:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.119.240;branch=z9hG4bKac1595438338;alias;received=192.168.119.240
From: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
To: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:221119966@192.168.119.227;transport=TCP>
Content-Length: 0


<------------>
[Jan 23 13:30:02] DEBUG[3414]: chan_sip.c:3604 __sip_xmit: Trying to put 'SIP/2.0 180' onto TCP socket destined for 192.168.119.240:5060
[Jan 23 13:30:05] DEBUG[3378]: chan_sip.c:4036 __sip_autodestruct: Auto destroying SIP dialog '207566505121200020742@192.168.165.33'
[Jan 23 13:30:05] DEBUG[3378]: chan_sip.c:5862 sip_destroy: Destroying SIP dialog 207566505121200020742@192.168.165.33
Really destroying SIP dialog '207566505121200020742@192.168.165.33' Method: REGISTER

<--- SIP read from UDP:192.168.165.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.119.227:5060;branch=z9hG4bK61f53d2d;rport
From: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
To: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 102 INVITE
Contact: <sip:221119966@192.168.165.33:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Type: application/sdp
Content-Length: 280

v=0
o=AudiocodesGW 1043900447 1043900313 IN IP4 192.168.165.33
s=Phone-Call
c=IN IP4 192.168.165.33
t=0 0
m=audio 30000 RTP/AVP 18 101
c=IN IP4 192.168.165.33
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 13 lines) ---
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:4116 __sip_ack: Acked pending invite 102
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:4153 __sip_ack: Stopping retransmission on '5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227' of Request 102: Match Found
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP o=AudiocodesGW 1043900447 1043900313 IN IP4 192.168.165.33... UNSUPPORTED.
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP s=Phone-Call... UNSUPPORTED.
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP c=IN IP4 192.168.165.33... OK.
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.
Found RTP audio format 18
Found RTP audio format 101
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP c=IN IP4 192.168.165.33... OK.
Found audio description format G729 for ID 18
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
Found audio description format telephone-event for ID 101
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK.
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.165.33:30000
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8696 process_sdp: Peer doesn't provide T.38 UDPTL
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:8706 process_sdp: We're settling with these formats: 0x100 (g729)
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:5715 update_call_counter: Updating call counter for outgoing call
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:12715 build_route: build_route: Contact hop: <sip:221119966@192.168.165.33:5060>
list_route: hop: <sip:221119966@192.168.165.33:5060>
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:9648 reqprep: Strict routing enforced for session 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
set_destination: Parsing <sip:221119966@192.168.165.33:5060> for address/port to send to
set_destination: set destination to 192.168.165.33, port 5060
Transmitting (no NAT) to 192.168.165.33:5060:
ACK sip:221119966@192.168.165.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.119.227:5060;branch=z9hG4bK57354afa;rport
Max-Forwards: 70
From: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
To: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
Contact: <sip:221119923@192.168.119.227>
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.16.1
Content-Length: 0


---
[Jan 23 13:30:07] DEBUG[3378]: chan_sip.c:3604 __sip_xmit: Trying to put 'ACK sip:223' onto UDP socket destined for 192.168.165.33:5060
    -- SIP/221119966-00000001 answered SIP/gateway-vg2-00000000
[Jan 23 13:30:07] DEBUG[3414]: chan_sip.c:25701 sip_set_rtp_peer: Early remote bridge setting SIP '1595423211231201113301@192.168.119.240' - Sending media to 192.168.165.33
[Jan 23 13:30:07] DEBUG[3414]: rtp.c:2196 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/gateway-vg2-00000000' with that of 'SIP/221119966-00000001'
[Jan 23 13:30:07] DEBUG[3414]: chan_sip.c:6271 sip_answer: SIP answering channel: SIP/gateway-vg2-00000000
[Jan 23 13:30:07] DEBUG[3414]: rtp.c:2688 ast_rtp_new_source: Setting the marker bit due to a source update
[Jan 23 13:30:07] DEBUG[3414]: chan_sip.c:10594 transmit_response_with_sdp: Setting framing from config on incoming call
[Jan 23 13:30:07] DEBUG[3414]: chan_sip.c:10291 add_sdp: ** Our capability: 0x108 (alaw|g729) Video flag: True Text flag: True
[Jan 23 13:30:07] DEBUG[3414]: chan_sip.c:10292 add_sdp: ** Our prefcodec: 0x0 (nothing)
Audio is at 192.168.119.227 port 11784
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Jan 23 13:30:07] DEBUG[3414]: chan_sip.c:10403 add_sdp: -- Done with adding codecs to SDP
[Jan 23 13:30:07] DEBUG[3414]: chan_sip.c:10527 add_sdp: Done building SDP. Settling with this capability: 0x108 (alaw|g729)

<--- Reliably Transmitting (no NAT) to 192.168.119.240:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.119.240;branch=z9hG4bKac1595438338;alias;received=192.168.119.240
From: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
To: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:221119966@192.168.119.227;transport=TCP>
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 256590941 256590941 IN IP4 192.168.165.33
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 192.168.165.33
t=0 0
m=audio 30000 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Jan 23 13:30:07] DEBUG[3414]: chan_sip.c:3604 __sip_xmit: Trying to put 'SIP/2.0 200' onto TCP socket destined for 192.168.119.240:5060
[Jan 23 13:30:07] DEBUG[3414]: rtp.c:2688 ast_rtp_new_source: Setting the marker bit due to a source update
[Jan 23 13:30:07] DEBUG[3414]: rtp.c:2688 ast_rtp_new_source: Setting the marker bit due to a source update
    -- Native bridging SIP/gateway-vg2-00000000 and SIP/221119966-00000001
[Jan 23 13:30:07] DEBUG[3350]: chan_sip.c:23302 sip_devicestate: Checking device state for peer 221119966
[Jan 23 13:30:07] DEBUG[3350]: devicestate.c:462 do_state_change: Changing state for SIP/221119966 - state 1 (Not in use)
[Jan 23 13:30:07] DEBUG[3350]: devicestate.c:442 devstate_event: device 'SIP/221119966' state '1'
[Jan 23 13:30:07] DEBUG[3350]: chan_sip.c:23302 sip_devicestate: Checking device state for peer gateway-vg2
[Jan 23 13:30:07] DEBUG[3350]: devicestate.c:462 do_state_change: Changing state for SIP/gateway-vg2 - state 1 (Not in use)
[Jan 23 13:30:07] DEBUG[3350]: devicestate.c:442 devstate_event: device 'SIP/gateway-vg2' state '1'
[Jan 23 13:30:07] DEBUG[3379]: app_queue.c:1092 handle_statechange: Device 'SIP/221119966' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Jan 23 13:30:07] DEBUG[3379]: app_queue.c:1092 handle_statechange: Device 'SIP/gateway-vg2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

<--- SIP read from TCP:192.168.119.240:65306 --->
ACK sip:221119966@192.168.119.227;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 192.168.119.240;branch=z9hG4bKac1683272124;alias
Max-Forwards: 70
From: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
To: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 1 ACK
Contact: <sip:221119923@192.168.119.240>
Supported: em,timer,replaces,path,histinfo,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.012.005
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
[Jan 23 13:30:07] DEBUG[3413]: chan_sip.c:4153 __sip_ack: Stopping retransmission on '1595423211231201113301@192.168.119.240' of Response 1: Match Not Found

<--- SIP read from UDP:192.168.165.33:5060 --->
INVITE sip:221119923@192.168.119.227 SIP/2.0
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac1071431721
Max-Forwards: 70
From: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
To: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 1 INVITE
Contact: <sip:221119966@192.168.165.33:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Type: application/sdp
Content-Length: 316

v=0
o=AudiocodesGW 1043900447 1043900314 IN IP4 192.168.165.33
s=Phone-Call
c=IN IP4 192.168.165.33
t=0 0
m=image 30002 udptl t38
c=IN IP4 192.168.165.33
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
--- (13 headers 13 lines) ---
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3418 parse_sip_options: Begin: parsing SIP "Supported: em,timer,replaces,path,early-session,resource-priority"
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3426 parse_sip_options: Found SIP option: -em-
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3448 parse_sip_options: Found no match for SIP option: em (Please file bug report!)
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3426 parse_sip_options: Found SIP option: -timer-
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3432 parse_sip_options: Matched SIP option: timer
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3426 parse_sip_options: Found SIP option: -replaces-
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3432 parse_sip_options: Matched SIP option: replaces
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3426 parse_sip_options: Found SIP option: -path-
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3432 parse_sip_options: Matched SIP option: path
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3426 parse_sip_options: Found SIP option: -early-session-
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3432 parse_sip_options: Matched SIP option: early-session
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3426 parse_sip_options: Found SIP option: -resource-priority-
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3432 parse_sip_options: Matched SIP option: resource-priority
Sending to 192.168.165.33 : 5060 (no NAT)
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:20183 handle_request_invite: Initializing initreq for method INVITE - callid 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP o=AudiocodesGW 1043900447 1043900314 IN IP4 192.168.165.33... UNSUPPORTED.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP s=Phone-Call... UNSUPPORTED.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP c=IN IP4 192.168.165.33... OK.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8334 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.
Got T.38 offer in SDP in dialog 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP c=IN IP4 192.168.165.33... OK.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:9073 process_sdp_a_image: FaxVersion: 0
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxVersion:0... OK.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:9050 process_sdp_a_image: T38MaxBitRate: 9600
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38MaxBitRate:9600... OK.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:9047 process_sdp_a_image: MaxBufferSize:1024
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxMaxBuffer:1024... OK.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:9082 process_sdp_a_image: FaxMaxDatagram: 400
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxMaxDatagram:400... OK.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:9119 process_sdp_a_image: RateManagement: transferredTCF
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxRateManagement:transferredTCF... OK.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:9126 process_sdp_a_image: UDP EC: t38UDPRedundancy
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxUdpEC:t38UDPRedundancy... OK.
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8658 process_sdp: Peer T.38 UDPTL is at port 192.168.165.33:30002
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:5102 change_t38_state: T38 state changed to 2 on channel SIP/221119966-00000001
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:8701 process_sdp: Have T.38 but no audio, accepting offer anyway
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:20348 handle_request_invite: Got a SIP re-invite for call 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:20608 handle_request_invite: SIP/221119966-00000001: This call is UP....

<--- Transmitting (no NAT) to 192.168.165.33:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac1071431721;received=192.168.165.33
From: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
To: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:221119923@192.168.119.227>
Content-Length: 0


<------------>
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:3604 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.165.33:5060
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:5102 change_t38_state: T38 state changed to 1 on channel SIP/gateway-vg2-00000000
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:9648 reqprep: Strict routing enforced for session 1595423211231201113301@192.168.119.240
set_destination: Parsing <sip:221119923@192.168.119.240> for address/port to send to
set_destination: set destination to 192.168.119.240, port 5060
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:10439 add_sdp: T.38 UDPTL is at 192.168.119.227 port 4052
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:10527 add_sdp: Done building SDP. Settling with this capability: 0x0 (nothing)
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:3288 initialize_initreq: Initializing already initialized SIP dialog 1595423211231201113301@192.168.119.240 (presumably reinvite)
Reliably Transmitting (no NAT) to 192.168.119.240:5060:
INVITE sip:221119923@192.168.119.240 SIP/2.0
Via: SIP/2.0/TCP 192.168.119.227:5060;branch=z9hG4bK60022403;rport
Max-Forwards: 70
rom: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
To: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
Contact: <sip:221119966@192.168.119.227;transport=TCP>
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 256590941 256590942 IN IP4 192.168.165.33
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 192.168.165.33
t=0 0
m=image 4052 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:204
a=T38FaxUdpEC:t38UDPFEC

---
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:3604 __sip_xmit: Trying to put 'INVITE sip:' onto TCP socket destined for 192.168.119.240:5060
[Jan 23 13:30:13] DEBUG[3414]: rtp.c:2688 ast_rtp_new_source: Setting the marker bit due to a source update

<--- SIP read from TCP:192.168.119.240:65306 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.119.227:5060;branch=z9hG4bK60022403;rport
From: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
To: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 102 INVITE
Contact: <sip:221119923@192.168.119.240>
Supported: em,timer,replaces,path,histinfo,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.012.005
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 325

v=0
o=AudiocodesGW 1595405269 1595404953 IN IP4 192.168.119.240
s=Phone-Call
c=IN IP4 192.168.119.240
t=0 0
m=image 10972 udptl t38
c=IN IP4 192.168.119.240
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
--- (13 headers 13 lines) ---
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP o=AudiocodesGW 1595405269 1595404953 IN IP4 192.168.119.240... UNSUPPORTED.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP s=Phone-Call... UNSUPPORTED.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP c=IN IP4 192.168.119.240... OK.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.
Got T.38 offer in SDP in dialog 1595423211231201113301@192.168.119.240
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP c=IN IP4 192.168.119.240... OK.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:9073 process_sdp_a_image: FaxVersion: 0
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxVersion:0... OK.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:9050 process_sdp_a_image: T38MaxBitRate: 9600
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38MaxBitRate:9600... OK.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:9047 process_sdp_a_image: MaxBufferSize:1024
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxMaxBuffer:1024... OK.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:9082 process_sdp_a_image: FaxMaxDatagram: 400
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxMaxDatagram:400... OK.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:9119 process_sdp_a_image: RateManagement: transferredTCF
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxRateManagement:transferredTCF... OK.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:9126 process_sdp_a_image: UDP EC: t38UDPRedundancy
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (image) SDP a=T38FaxUdpEC:t38UDPRedundancy... OK.
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8658 process_sdp: Peer T.38 UDPTL is at port 192.168.119.240:10972
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:5102 change_t38_state: T38 state changed to 3 on channel SIP/gateway-vg2-00000000
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:8701 process_sdp: Have T.38 but no audio, accepting offer anyway
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:5715 update_call_counter: Updating call counter for incoming call
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:9648 reqprep: Strict routing enforced for session 1595423211231201113301@192.168.119.240
set_destination: Parsing <sip:221119923@192.168.119.240> for address/port to send to
set_destination: set destination to 192.168.119.240, port 5060
Transmitting (no NAT) to 192.168.119.240:5060:
ACK sip:221119923@192.168.119.240 SIP/2.0
Via: SIP/2.0/TCP 192.168.119.227:5060;branch=z9hG4bK4d6cfb6e;rport
Max-Forwards: 70
From: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
To: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
Contact: <sip:221119966@192.168.119.227;transport=TCP>
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.16.1
Content-Length: 0


---
[Jan 23 13:30:13] DEBUG[3413]: chan_sip.c:3604 __sip_xmit: Trying to put 'ACK sip:223' onto TCP socket destined for 192.168.119.240:5060
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:5102 change_t38_state: T38 state changed to 3 on channel SIP/221119966-00000001
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:10439 add_sdp: T.38 UDPTL is at 192.168.119.227 port 4649
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:10527 add_sdp: Done building SDP. Settling with this capability: 0x0 (nothing)

<--- Reliably Transmitting (no NAT) to 192.168.165.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac1071431721;received=192.168.165.33
From: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
To: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:221119923@192.168.119.227>
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 1581035008 1581035009 IN IP4 192.168.119.240
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 192.168.119.240
t=0 0
m=image 4649 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:397
a=T38FaxUdpEC:t38UDPRedundancy

<------------>
[Jan 23 13:30:13] DEBUG[3414]: chan_sip.c:3604 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.165.33:5060

<--- SIP read from UDP:192.168.165.33:5060 --->
ACK sip:221119923@192.168.119.227 SIP/2.0
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac1071941224
Max-Forwards: 70
From: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
To: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 1 ACK
Contact: <sip:221119966@192.168.165.33:5060>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
[Jan 23 13:30:13] DEBUG[3378]: chan_sip.c:4153 __sip_ack: Stopping retransmission on '5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227' of Response 1: Match Found

<--- SIP read from UDP:192.168.165.33:5060 --->
BYE sip:221119923@192.168.119.227 SIP/2.0
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac1175380438
Max-Forwards: 70
From: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
To: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 2 BYE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Reason: Q.850 ;cause=16 ;text="local"
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
[Jan 23 13:30:54] DEBUG[3378]: chan_sip.c:21306 handle_request_bye: Initializing initreq for method BYE - callid 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
Sending to 192.168.165.33 : 5060 (no NAT)
[Jan 23 13:30:54] DEBUG[3378]: chan_sip.c:3301 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
[Jan 23 13:30:54] DEBUG[3378]: chan_sip.c:21404 handle_request_bye: Received bye, issuing owner hangup

<--- Transmitting (no NAT) to 192.168.165.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.165.33;branch=z9hG4bKac1175380438;received=192.168.165.33
From: <sip:221119966@192.168.165.33:5060>;tag=1c1043871951
To: "Wywolanie" <sip:221119923@192.168.119.227>;tag=as78ed8fa4
Call-ID: 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
CSeq: 2 BYE
Server: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Jan 23 13:30:54] DEBUG[3378]: chan_sip.c:3604 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.165.33:5060
[Jan 23 13:30:54] DEBUG[3414]: rtp.c:4106 bridge_native_loop: Oooh, got a hangup
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:25703 sip_set_rtp_peer: Sending reinvite on SIP '1595423211231201113301@192.168.119.240' - It's audio soon redirected to IP 192.168.119.227
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:9648 reqprep: Strict routing enforced for session 1595423211231201113301@192.168.119.240
set_destination: Parsing <sip:221119923@192.168.119.240> for address/port to send to
set_destination: set destination to 192.168.119.240, port 5060
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:10291 add_sdp: ** Our capability: 0x108 (alaw|g729) Video flag: True Text flag: True
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:10292 add_sdp: ** Our prefcodec: 0x0 (nothing)
Audio is at 192.168.119.227 port 11784
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:10403 add_sdp: -- Done with adding codecs to SDP
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:10527 add_sdp: Done building SDP. Settling with this capability: 0x108 (alaw|g729)
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:3288 initialize_initreq: Initializing already initialized SIP dialog 1595423211231201113301@192.168.119.240 (presumably reinvite)
Reliably Transmitting (no NAT) to 192.168.119.240:5060:
INVITE sip:221119923@192.168.119.240 SIP/2.0
Via: SIP/2.0/TCP 192.168.119.227:5060;branch=z9hG4bK2bb4269d;rport
Max-Forwards: 70
From: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
To: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
Contact: <sip:221119966@192.168.119.227;transport=TCP>
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.16.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 256590941 256590943 IN IP4 192.168.119.227
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 192.168.119.227
t=0 0
m=audio 11784 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:3604 __sip_xmit: Trying to put 'INVITE sip:' onto TCP socket destined for 192.168.119.240:5060
[Jan 23 13:30:54] DEBUG[3414]: channel.c:5556 ast_channel_bridge: Returning from native bridge, channels: SIP/gateway-vg2-00000000, SIP/221119966-00000001
[Jan 23 13:30:54] DEBUG[3414]: channel.c:1855 ast_hangup: Hanging up channel 'SIP/221119966-00000001'
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:6080 sip_hangup: Hangup call SIP/221119966-00000001, SIP callid 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
[Jan 23 13:30:54] DEBUG[3414]: rtp.c:2146 ast_rtp_early_bridge: Channel '<unspecified>' has no RTP, not doing anything
[Jan 23 13:30:54] DEBUG[3350]: chan_sip.c:23302 sip_devicestate: Checking device state for peer 221119966
[Jan 23 13:30:54] DEBUG[3350]: devicestate.c:462 do_state_change: Changing state for SIP/221119966 - state 1 (Not in use)
[Jan 23 13:30:54] DEBUG[3350]: devicestate.c:442 devstate_event: device 'SIP/221119966' state '1'
[Jan 23 13:30:54] DEBUG[3379]: app_queue.c:1092 handle_statechange: Device 'SIP/221119966' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Jan 23 13:30:54] DEBUG[3414]: app_dial.c:2335 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Jan 23 13:30:54] DEBUG[3414]: pbx.c:4361 __ast_pbx_run: Spawn extension (xxx_test_in,221119966,1) exited non-zero on 'SIP/gateway-vg2-00000000'
  == Spawn extension (xxx_test_in, 221119966, 1) exited non-zero on 'SIP/gateway-vg2-00000000'
[Jan 23 13:30:54] DEBUG[3414]: channel.c:1750 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/gateway-vg2-00000000'
[Jan 23 13:30:54] DEBUG[3414]: channel.c:1855 ast_hangup: Hanging up channel 'SIP/gateway-vg2-00000000'
[Jan 23 13:30:54] DEBUG[3414]: chan_sip.c:6080 sip_hangup: Hangup call SIP/gateway-vg2-00000000, SIP callid 1595423211231201113301@192.168.119.240
Scheduling destruction of SIP dialog '1595423211231201113301@192.168.119.240' in 32000 ms (Method: INVITE)

<--- SIP read from TCP:192.168.119.240:65306 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.119.227:5060;branch=z9hG4bK2bb4269d;rport
From: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
To: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 103 INVITE
Contact: <sip:221119923@192.168.119.240>
Supported: em,timer,replaces,path,histinfo,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.012.005
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 289

v=0
o=AudiocodesGW 1595405269 1595404954 IN IP4 192.168.119.240
s=Phone-Call
c=IN IP4 192.168.119.240
t=0 0
m=audio 10970 RTP/AVP 18 101
c=IN IP4 192.168.119.240
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
[Jan 23 13:30:54] DEBUG[3350]: chan_sip.c:23302 sip_devicestate: Checking device state for peer gateway-vg2
[Jan 23 13:30:54] DEBUG[3350]: devicestate.c:462 do_state_change: Changing state for SIP/gateway-vg2 - state 1 (Not in use)
[Jan 23 13:30:54] DEBUG[3350]: devicestate.c:442 devstate_event: device 'SIP/gateway-vg2' state '1'
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP o=AudiocodesGW 1595405269 1595404954 IN IP4 192.168.119.240... UNSUPPORTED.
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP s=Phone-Call... UNSUPPORTED.
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP c=IN IP4 192.168.119.240... OK.
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8334 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.
Found RTP audio format 18
Found RTP audio format 101
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP c=IN IP4 192.168.119.240... OK.
Found audio description format G729 for ID 18
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
Found audio description format telephone-event for ID 101
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=ptime:20... OK.
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8502 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:5102 change_t38_state: T38 state changed to 0 on channel <none>
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.119.240:10970
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8696 process_sdp: Peer doesn't provide T.38 UDPTL
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:8706 process_sdp: We're settling with these formats: 0x100 (g729)
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:5715 update_call_counter: Updating call counter for incoming call
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:12655 build_route: build_route: Retaining previous route: <sip:221119923@192.168.119.240>
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:9648 reqprep: Strict routing enforced for session 1595423211231201113301@192.168.119.240
set_destination: Parsing <sip:221119923@192.168.119.240> for address/port to send to
set_destination: set destination to 192.168.119.240, port 5060
Transmitting (no NAT) to 192.168.119.240:5060:
ACK sip:221119923@192.168.119.240 SIP/2.0
Via: SIP/2.0/TCP 192.168.119.227:5060;branch=z9hG4bK1338fd82;rport
Max-Forwards: 70
From: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
To: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
Contact: <sip:221119966@192.168.119.227;transport=TCP>
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.16.1
Content-Length: 0


---
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:3604 __sip_xmit: Trying to put 'ACK sip:223' onto TCP socket destined for 192.168.119.240:5060
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:9648 reqprep: Strict routing enforced for session 1595423211231201113301@192.168.119.240
set_destination: Parsing <sip:221119923@192.168.119.240> for address/port to send to
set_destination: set destination to 192.168.119.240, port 5060
Reliably Transmitting (no NAT) to 192.168.119.240:5060:
BYE sip:221119923@192.168.119.240 SIP/2.0
Via: SIP/2.0/TCP 192.168.119.227:5060;branch=z9hG4bK7f9be90c;rport
Max-Forwards: 70
From: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
To: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.16.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Jan 23 13:30:54] DEBUG[3413]: chan_sip.c:3604 __sip_xmit: Trying to put 'BYE sip:223' onto TCP socket destined for 192.168.119.240:5060
Scheduling destruction of SIP dialog '1595423211231201113301@192.168.119.240' in 32000 ms (Method: INVITE)
[Jan 23 13:30:54] DEBUG[3379]: app_queue.c:1092 handle_statechange: Device 'SIP/gateway-vg2' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

<--- SIP read from TCP:192.168.119.240:65306 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.119.227:5060;branch=z9hG4bK7f9be90c;rport
From: <sip:221119966@192.168.119.227;user=phone>;tag=as645d2fea
To: "Wywolanie" <sip:221119923@audiocodes.com>;tag=1c1595424280
Call-ID: 1595423211231201113301@192.168.119.240
CSeq: 104 BYE
Contact: <sip:221119923@192.168.119.240>
Supported: em,timer,replaces,path,histinfo,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.012.005
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
[Jan 23 13:30:55] DEBUG[3378]: chan_sip.c:5862 sip_destroy: Destroying SIP dialog 1595423211231201113301@192.168.119.240
Really destroying SIP dialog '1595423211231201113301@192.168.119.240' Method: INVITE
[Jan 23 13:30:55] DEBUG[3378]: chan_sip.c:5862 sip_destroy: Destroying SIP dialog 5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227
Really destroying SIP dialog '5bf4ac1d39f5d41a02cd0ee306fbd4ba@192.168.119.227' Method: BYE
obelix*CLI>
obelix*CLI> exit
Executing last minute cleanups
Asterisk ending (0).
obelix:~#