<-- SIP read from 192.168.1.173:5060:
INVITE sip:0315400401@192.168.1.153 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.173;branch=z9hG4bK626bffff94efffff
From: "0315400504" <sip:0315400504@192.168.1.153>;tag=e7aaffff29850000
To: <sip:0315400401@192.168.1.153>
Contact: <sip:0315400504@192.168.1.173>
Supported: replaces
Call-ID: 32810000248dffff@192.168.1.173
CSeq: 13798 INVITE
User-Agent: Grandstream HT496 1.0.2.6
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 366

v=0
o=0315400504 8000 8000 IN IP4 192.168.1.173
s=SIP Call
c=IN IP4 192.168.1.173
t=0 0
m=audio 5004 RTP/AVP 18 4 0 2 15 97 96
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-11

[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 0: INVITE sip:0315400401@192.168.1.153 SIP/2.0 (44)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.173;branch=z9hG4bK626bffff94efffff (62)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 2: From: "0315400504" <sip:0315400504@192.168.1.153>;tag=e7aaffff29850000 (71)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 3: To: <sip:0315400401@192.168.1.153> (35)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 4: Contact: <sip:0315400504@192.168.1.173> (40)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 5: Supported: replaces (19)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 6: Call-ID: 32810000248dffff@192.168.1.173 (40)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 7: CSeq: 13798 INVITE (18)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 8: User-Agent: Grandstream HT496 1.0.2.6 (37)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 9: Max-Forwards: 70 (16)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 11: Content-Type: application/sdp (29)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 12: Content-Length: 366 (19)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 13:  (0)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: v=0 (3)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: o=0315400504 8000 8000 IN IP4 192.168.1.173 (44)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: s=SIP Call (10)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: c=IN IP4 192.168.1.173 (23)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: t=0 0 (5)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: m=audio 5004 RTP/AVP 18 4 0 2 15 97 96 (38)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=sendrecv (10)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:18 G729/8000 (21)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:4 G723/8000 (20)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:2 G726-32/8000 (23)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:15 G728/8000 (21)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:97 iLBC/8000 (21)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=fmtp:97 mode=20 (17)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=ptime:20 (10)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:96 telephone-event/8000 (32)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=fmtp:96 0-11 (14)
--- (13 headers 17 lines)---
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4109 sip_alloc: Allocating new SIP dialog for 32810000248dffff@192.168.1.173 - INVITE (With RTP)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:14007 handle_request: **** Received INVITE (5) - Command in SIP INVITE
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:1611 parse_sip_options: Begin: parsing SIP "Supported: replaces"
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:1619 parse_sip_options: Found SIP option: -replaces-
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:1625 parse_sip_options: Matched SIP option: replaces
Sending to 192.168.1.173 : 5060 (no NAT)
Using INVITE request as basis request - 32810000248dffff@192.168.1.173
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:8649 check_user_full: Setting NAT on RTP to Off
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:8659 check_user_full: Setting NAT on UDPTL to Off
Found user '0315400504'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 97
Found RTP audio format 96
Peer audio RTP is at port 192.168.1.173:5004
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4648 process_sdp: Peer doesn't provide T.38 UDPTL
Found description format G729 for ID 18
Found description format G723 for ID 4
Found description format PCMU for ID 0
Found description format G726-32 for ID 2
Found description format G728 for ID 15
Found description format iLBC for ID 97
Got unsupported a:fmtp in SDP offer
Got unsupported a:ptime in SDP offer
Found description format telephone-event for ID 96
Got unsupported a:fmtp in SDP offer
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4834 process_sdp: T38 state changed to 0 on channel <none>
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd05 (g723|ulaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.173:5004
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4904 process_sdp: We're settling with these formats: 0x4 (ulaw)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:12820 handle_request_invite: Checking SIP call limits for device 0315400504
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:2870 update_call_counter: Updating call counter for incoming call
Looking for 0315400401 in centre (domain 192.168.1.153)
[Aug 30 09:03:13] DEBUG[26104]: frame.c:1037 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:3606 sip_new: *** Our native formats are 0x4 (ulaw)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:3607 sip_new: *** Joint capabilities are 0x4 (ulaw)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:3608 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263)
[Aug 30 09:03:13] DEBUG[26104]: frame.c:1037 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:3609 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:3632 sip_new: This channel will not be able to handle video.
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:7590 build_route: build_route: Contact hop: <sip:0315400504@192.168.1.173>
list_route: hop: <sip:0315400504@192.168.1.173>
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:12890 handle_request_invite: SIP/0315400504-007a1250: New call is still down.... Trying...
Transmitting (no NAT) to 192.168.1.173:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.173;branch=z9hG4bK626bffff94efffff;received=192.168.1.173
From: "0315400504" <sip:0315400504@192.168.1.153>;tag=e7aaffff29850000
To: <sip:0315400401@192.168.1.153>
Call-ID: 32810000248dffff@192.168.1.173
CSeq: 13798 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0315400401@192.168.1.153>
Content-Length: 0


---
[Aug 30 09:03:13] DEBUG[26104]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/0315400504-007a1250
[Aug 30 09:03:13] DEBUG[26079]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 0315400504
[Aug 30 09:03:13] DEBUG[26079]: chan_sip.c:14591 sip_devicestate: Checking device state for peer 0315400504
[Aug 30 09:03:13] DEBUG[26112]: pbx.c:1680 pbx_extension_helper: Launching 'Goto'
[Aug 30 09:03:13] DEBUG[26079]: devicestate.c:287 do_state_change: Changing state for SIP/0315400504 - state 1 (Not in use)
    -- Executing [0315400401@centre:1] Goto("SIP/0315400504-007a1250", "all|0315400401|1") in new stack
[Aug 30 09:03:13] DEBUG[26113]: app_queue.c:533 changethread: Device 'SIP/0315400504' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
    -- Goto (all,0315400401,1)
[Aug 30 09:03:13] DEBUG[26112]: pbx.c:1680 pbx_extension_helper: Launching 'Dial'
    -- Executing [0315400401@all:1] Dial("SIP/0315400504-007a1250", "SIP/ssw/0315400401") in new stack
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:14649 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4109 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:2501 create_addr_from_peer: Our T38 capability (3872)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:2511 create_addr_from_peer: Setting NAT on RTP to Off
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:2523 create_addr_from_peer: Setting NAT on UDPTL to Off
[Aug 30 09:03:13] DEBUG[26112]: frame.c:1037 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:3606 sip_new: *** Our native formats are 0x4 (ulaw)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:3607 sip_new: *** Joint capabilities are 0x0 (nothing)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:3608 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263)
[Aug 30 09:03:13] DEBUG[26112]: frame.c:1037 ast_codec_choose: Could not find preferred codec - Going for the best codec
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:3609 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:3611 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:3632 sip_new: This channel will not be able to handle video.
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:16144 sip_set_rtp_peer: Early remote bridge setting SIP '106aec5b00b8145d46db39c050c4d425@192.168.1.153' - Sending media to 192.168.1.173
[Aug 30 09:03:13] DEBUG[26112]: rtp.c:1486 ast_rtp_make_compatible: Seeded SDP of 'SIP/ssw-007b4d30' with that of 'SIP/0315400504-007a1250'
[Aug 30 09:03:13] DEBUG[26112]: channel.c:3022 ast_channel_inherit_variables: Not copying variable STACK-all-0315400401-1.
[Aug 30 09:03:13] DEBUG[26112]: channel.c:3022 ast_channel_inherit_variables: Not copying variable STACK-centre-0315400401-1.
[Aug 30 09:03:13] DEBUG[26112]: channel.c:3022 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Aug 30 09:03:13] DEBUG[26112]: channel.c:3022 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT.
[Aug 30 09:03:13] DEBUG[26112]: channel.c:3022 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Aug 30 09:03:13] DEBUG[26112]: channel.c:3022 ast_channel_inherit_variables: Not copying variable SIPURI.
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:2709 sip_call: Outgoing Call for 0315400401
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:2870 update_call_counter: Updating call counter for outgoing call
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:2717 sip_call: Our T38 capability (3872), joint T38 capability (3872)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:5806 add_sdp: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:5807 add_sdp: ** Our prefcodec: 0x4 (ulaw)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:5822 add_sdp: This call needs video offers, but there's no video support enabled !
Audio is at 192.168.1.153 port 15988
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:5959 add_sdp: -- Done with adding codecs to SDP
[Aug 30 09:03:13] DEBUG[26112]: channel.c:2156 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:5998 add_sdp: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 0: INVITE sip:0315400401@192.168.1.5 SIP/2.0 (42)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.153:5060;branch=z9hG4bK375a8764;rport (65)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 2: From: "0315400504" <sip:0315400504@192.168.1.153>;tag=as7eb208ae (65)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 3: To: <sip:0315400401@192.168.1.5> (33)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 4: Contact: <sip:0315400504@192.168.1.153> (40)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 5: Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 (56)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 6: CSeq: 102 INVITE (16)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 7: User-Agent: Asterisk PBX (24)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 8: Max-Forwards: 70 (16)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 9: Date: Wed, 30 Aug 2006 06:03:13 GMT (35)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 11: Supported: replaces (19)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 12: Content-Type: application/sdp (29)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 13: Content-Length: 275 (19)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4329 parse_request: Header 14:  (0)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: v=0 (3)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: o=root 26076 26076 IN IP4 192.168.1.173 (40)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: s=session (9)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: c=IN IP4 192.168.1.173 (23)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: t=0 0 (5)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: m=audio 5004 RTP/AVP 0 3 8 96 (29)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: a=rtpmap:0 PCMU/8000 (20)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: a=rtpmap:3 GSM/8000 (19)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: a=rtpmap:8 PCMA/8000 (20)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: a=rtpmap:96 telephone-event/8000 (32)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: a=fmtp:96 0-16 (14)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: a=silenceSupp:off - - - - (25)
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:4361 parse_request: Line: a=sendrecv (10)
Reliably Transmitting (no NAT) to 192.168.1.5:5060:
INVITE sip:0315400401@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.153:5060;branch=z9hG4bK375a8764;rport
From: "0315400504" <sip:0315400504@192.168.1.153>;tag=as7eb208ae
To: <sip:0315400401@192.168.1.5>
Contact: <sip:0315400504@192.168.1.153>
Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 30 Aug 2006 06:03:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 275

v=0
o=root 26076 26076 IN IP4 192.168.1.173
s=session
c=IN IP4 192.168.1.173
t=0 0
m=audio 5004 RTP/AVP 0 3 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
[Aug 30 09:03:13] DEBUG[26112]: chan_sip.c:1902 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #17
    -- Called ssw/0315400401

<-- SIP read from 192.168.1.5:5060:
SIP/2.0 100 Trying
From: "0315400504"<sip:0315400504@192.168.1.153>;tag=as7eb208ae
To: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9
Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.153:5060;rport=5060;branch=z9hG4bK375a8764
Contact: <sip:0315400401@192.168.1.5>
Content-Length: 0


[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 0: SIP/2.0 100 Trying (18)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 1: From: "0315400504"<sip:0315400504@192.168.1.153>;tag=as7eb208ae (64)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 2: To: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9 (73)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 3: Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 (56)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 4: CSeq: 102 INVITE (16)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.1.153:5060;rport=5060;branch=z9hG4bK375a8764 (70)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 6: Contact: <sip:0315400401@192.168.1.5> (38)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 7: Content-Length: 0 (17)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 8:  (0)
--- (8 headers 0 lines)---
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:4157 find_call: = Found Their Call ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 Their Tag  Our tag: as7eb208ae
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:2038 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #17 - INVITE (got response)
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:2047 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '106aec5b00b8145d46db39c050c4d425@192.168.1.153' Request 102: Found
[Aug 30 09:03:13] DEBUG[26104]: chan_sip.c:11178 handle_response_invite: SIP response 100 to standard invite

<-- SIP read from 192.168.1.5:5060:
SIP/2.0 183 Session Progress
From: "0315400504"<sip:0315400504@192.168.1.153>;tag=as7eb208ae
To: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9
Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.153:5060;rport=5060;branch=z9hG4bK375a8764
Contact: <sip:0315400401@192.168.1.5>
Content-Type: application/SDP
Content-Length: 230

v=0
o=AudiocodesGW 689998298 689998178 IN IP4 192.168.1.38
s=Phone-Call
c=IN IP4 192.168.1.38
t=0 0
m=audio 6000 RTP/AVP 0 96
a=rtpmap:0 pcmu/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv

[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 0: SIP/2.0 183 Session Progress (28)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 1: From: "0315400504"<sip:0315400504@192.168.1.153>;tag=as7eb208ae (64)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 2: To: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9 (73)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 3: Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 (56)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 4: CSeq: 102 INVITE (16)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.1.153:5060;rport=5060;branch=z9hG4bK375a8764 (70)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 6: Contact: <sip:0315400401@192.168.1.5> (38)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 7: Content-Type: application/SDP (29)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 8: Content-Length: 230 (19)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 9:  (0)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: v=0 (3)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: o=AudiocodesGW 689998298 689998178 IN IP4 192.168.1.38 (55)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: s=Phone-Call (12)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: c=IN IP4 192.168.1.38 (22)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: t=0 0 (5)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: m=audio 6000 RTP/AVP 0 96 (25)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:0 pcmu/8000 (20)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:96 telephone-event/8000 (32)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=fmtp:96 0-15 (14)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=ptime:20 (10)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=sendrecv (10)
--- (9 headers 11 lines)---
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4157 find_call: = Found Their Call ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 Their Tag 5b16956-13c4-44f5390d-625f8523-32f9 Our tag: as7eb208ae
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:2047 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '106aec5b00b8145d46db39c050c4d425@192.168.1.153' Request 102: Found
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:11178 handle_response_invite: SIP response 183 to standard invite
Found RTP audio format 0
Found RTP audio format 96
Peer audio RTP is at port 192.168.1.38:6000
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4648 process_sdp: Peer doesn't provide T.38 UDPTL
Found description format pcmu for ID 0
Found description format telephone-event for ID 96
Got unsupported a:fmtp in SDP offer
Got unsupported a:ptime in SDP offer
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4834 process_sdp: T38 state changed to 0 on channel SIP/ssw-007b4d30
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.38:6000
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4904 process_sdp: We're settling with these formats: 0x4 (ulaw)
[Aug 30 09:03:14] DEBUG[26104]: chan_sip.c:4911 process_sdp: We have an owner, now see if we need to change this call
    -- SIP/ssw-007b4d30 is making progress passing it to SIP/0315400504-007a1250
[Aug 30 09:03:14] DEBUG[26112]: chan_sip.c:16144 sip_set_rtp_peer: Early remote bridge setting SIP '32810000248dffff@192.168.1.173' - Sending media to 192.168.1.38
[Aug 30 09:03:14] DEBUG[26112]: rtp.c:1421 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/0315400504-007a1250' with that of 'SIP/ssw-007b4d30'
[Aug 30 09:03:14] DEBUG[26112]: chan_sip.c:5806 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True
[Aug 30 09:03:14] DEBUG[26112]: chan_sip.c:5807 add_sdp: ** Our prefcodec: 0x0 (nothing)
Audio is at 192.168.1.153 port 12752
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Aug 30 09:03:14] DEBUG[26112]: chan_sip.c:5959 add_sdp: -- Done with adding codecs to SDP
[Aug 30 09:03:14] DEBUG[26112]: channel.c:2156 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1)
[Aug 30 09:03:14] DEBUG[26112]: chan_sip.c:5998 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw)
Transmitting (no NAT) to 192.168.1.173:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.173;branch=z9hG4bK626bffff94efffff;received=192.168.1.173
From: "0315400504" <sip:0315400504@192.168.1.153>;tag=e7aaffff29850000
To: <sip:0315400401@192.168.1.153>;tag=as443de35d
Call-ID: 32810000248dffff@192.168.1.173
CSeq: 13798 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0315400401@192.168.1.153>
Content-Type: application/sdp
Content-Length: 226

v=0
o=root 26076 26076 IN IP4 192.168.1.38
s=session
c=IN IP4 192.168.1.38
t=0 0
m=audio 6000 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=sendrecv

---

<-- SIP read from 192.168.1.173:5060:

--- (0 headers 0 lines) Nat keepalive ---

<-- SIP read from 192.168.1.5:5060:
SIP/2.0 200 OK
From: "0315400504"<sip:0315400504@192.168.1.153>;tag=as7eb208ae
To: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9
Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.153:5060;rport=5060;branch=z9hG4bK375a8764
Contact: <sip:0315400401@192.168.1.5>
Content-Type: application/SDP
Content-Length: 230

v=0
o=AudiocodesGW 689998298 689998178 IN IP4 192.168.1.38
s=Phone-Call
c=IN IP4 192.168.1.38
t=0 0
m=audio 6000 RTP/AVP 0 96
a=rtpmap:0 pcmu/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv

[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 0: SIP/2.0 200 OK (14)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 1: From: "0315400504"<sip:0315400504@192.168.1.153>;tag=as7eb208ae (64)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 2: To: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9 (73)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 3: Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 (56)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 4: CSeq: 102 INVITE (16)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.1.153:5060;rport=5060;branch=z9hG4bK375a8764 (70)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 6: Contact: <sip:0315400401@192.168.1.5> (38)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 7: Content-Type: application/SDP (29)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 8: Content-Length: 230 (19)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 9:  (0)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: v=0 (3)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: o=AudiocodesGW 689998298 689998178 IN IP4 192.168.1.38 (55)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: s=Phone-Call (12)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: c=IN IP4 192.168.1.38 (22)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: t=0 0 (5)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: m=audio 6000 RTP/AVP 0 96 (25)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:0 pcmu/8000 (20)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=rtpmap:96 telephone-event/8000 (32)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=fmtp:96 0-15 (14)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=ptime:20 (10)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=sendrecv (10)
--- (9 headers 11 lines)---
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4157 find_call: = Found Their Call ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 Their Tag 5b16956-13c4-44f5390d-625f8523-32f9 Our tag: as7eb208ae
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:1987 __sip_ack: Acked pending invite 102
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:2005 __sip_ack: Stopping retransmission on '106aec5b00b8145d46db39c050c4d425@192.168.1.153' of Request 102: Match Not Found
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:11178 handle_response_invite: SIP response 200 to standard invite
Found RTP audio format 0
Found RTP audio format 96
Peer audio RTP is at port 192.168.1.38:6000
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4648 process_sdp: Peer doesn't provide T.38 UDPTL
Found description format pcmu for ID 0
Found description format telephone-event for ID 96
Got unsupported a:fmtp in SDP offer
Got unsupported a:ptime in SDP offer
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4834 process_sdp: T38 state changed to 0 on channel SIP/ssw-007b4d30
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.38:6000
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4904 process_sdp: We're settling with these formats: 0x4 (ulaw)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4911 process_sdp: We have an owner, now see if we need to change this call
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:2870 update_call_counter: Updating call counter for outgoing call
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:7590 build_route: build_route: Contact hop: <sip:0315400401@192.168.1.5>
list_route: hop: <sip:0315400401@192.168.1.5>
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:5337 reqprep: Strict routing enforced for session 106aec5b00b8145d46db39c050c4d425@192.168.1.153
set_destination: Parsing <sip:0315400401@192.168.1.5> for address/port to send to
set_destination: set destination to 192.168.1.5, port 5060
Transmitting (no NAT) to 192.168.1.5:5060:
ACK sip:0315400401@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.153:5060;branch=z9hG4bK3a5b05a6;rport
From: "0315400504" <sip:0315400504@192.168.1.153>;tag=as7eb208ae
To: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9
Contact: <sip:0315400504@192.168.1.153>
Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Aug 30 09:03:19] DEBUG[26112]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/ssw-007b4d30
    -- SIP/ssw-007b4d30 answered SIP/0315400504-007a1250
[Aug 30 09:03:19] DEBUG[26112]: rtp.c:1421 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/0315400504-007a1250' with that of 'SIP/ssw-007b4d30'
[Aug 30 09:03:19] DEBUG[26079]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - ssw
[Aug 30 09:03:19] DEBUG[26079]: chan_sip.c:14591 sip_devicestate: Checking device state for peer ssw
[Aug 30 09:03:19] DEBUG[26079]: devicestate.c:287 do_state_change: Changing state for SIP/ssw - state 1 (Not in use)
[Aug 30 09:03:19] DEBUG[26114]: app_queue.c:533 changethread: Device 'SIP/ssw' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Aug 30 09:03:19] DEBUG[26112]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/0315400504-007a1250
[Aug 30 09:03:19] DEBUG[26112]: chan_sip.c:3310 sip_answer: SIP answering channel: SIP/0315400504-007a1250
[Aug 30 09:03:19] DEBUG[26079]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 0315400504
[Aug 30 09:03:19] DEBUG[26079]: chan_sip.c:14591 sip_devicestate: Checking device state for peer 0315400504
[Aug 30 09:03:19] DEBUG[26079]: devicestate.c:287 do_state_change: Changing state for SIP/0315400504 - state 1 (Not in use)
[Aug 30 09:03:19] DEBUG[26115]: app_queue.c:533 changethread: Device 'SIP/0315400504' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Aug 30 09:03:19] DEBUG[26112]: chan_sip.c:5806 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True
[Aug 30 09:03:19] DEBUG[26112]: chan_sip.c:5807 add_sdp: ** Our prefcodec: 0x0 (nothing)
Audio is at 192.168.1.153 port 12752
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Aug 30 09:03:19] DEBUG[26112]: chan_sip.c:5959 add_sdp: -- Done with adding codecs to SDP
[Aug 30 09:03:19] DEBUG[26112]: channel.c:2156 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1)
[Aug 30 09:03:19] DEBUG[26112]: chan_sip.c:5998 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw)
Reliably Transmitting (no NAT) to 192.168.1.173:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.173;branch=z9hG4bK626bffff94efffff;received=192.168.1.173
From: "0315400504" <sip:0315400504@192.168.1.153>;tag=e7aaffff29850000
To: <sip:0315400401@192.168.1.153>;tag=as443de35d
Call-ID: 32810000248dffff@192.168.1.173
CSeq: 13798 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0315400401@192.168.1.153>
Content-Type: application/sdp
Content-Length: 226

v=0
o=root 26076 26077 IN IP4 192.168.1.38
s=session
c=IN IP4 192.168.1.38
t=0 0
m=audio 6000 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
[Aug 30 09:03:19] DEBUG[26112]: chan_sip.c:1902 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #19
    -- Native bridging SIP/0315400504-007a1250 and SIP/ssw-007b4d30

<-- SIP read from 192.168.1.173:5060:
ACK sip:0315400401@192.168.1.153 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.173;branch=z9hG4bK1b36000023170000
From: "0315400504" <sip:0315400504@192.168.1.153>;tag=e7aaffff29850000
To: <sip:0315400401@192.168.1.153>;tag=as443de35d
Contact: <sip:0315400504@192.168.1.173>
Call-ID: 32810000248dffff@192.168.1.173
CSeq: 13798 ACK
User-Agent: Grandstream HT496 1.0.2.6
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 0: ACK sip:0315400401@192.168.1.153 SIP/2.0 (41)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.173;branch=z9hG4bK1b36000023170000 (62)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 2: From: "0315400504" <sip:0315400504@192.168.1.153>;tag=e7aaffff29850000 (71)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 3: To: <sip:0315400401@192.168.1.153>;tag=as443de35d (50)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 4: Contact: <sip:0315400504@192.168.1.173> (40)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 5: Call-ID: 32810000248dffff@192.168.1.173 (40)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 6: CSeq: 13798 ACK (15)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 7: User-Agent: Grandstream HT496 1.0.2.6 (37)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 8: Max-Forwards: 70 (16)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 10: Content-Length: 0 (17)
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 11:  (0)
--- (11 headers 0 lines)---
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4157 find_call: = No match Their Call ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 Their Tag 5b16956-13c4-44f5390d-625f8523-32f9 Our tag: as7eb208ae
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:4157 find_call: = Found Their Call ID: 32810000248dffff@192.168.1.173 Their Tag e7aaffff29850000 Our tag: as443de35d
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:14007 handle_request: **** Received ACK (6) - Command in SIP ACK
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:1995 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #19
[Aug 30 09:03:19] DEBUG[26104]: chan_sip.c:2005 __sip_ack: Stopping retransmission on '32810000248dffff@192.168.1.173' of Response 13798: Match Not Found

<-- SIP read from 192.168.1.5:5060:
INVITE sip:0315400504@192.168.1.153 SIP/2.0
From: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9
To: "0315400504"<sip:0315400504@192.168.1.153>;tag=as7eb208ae
Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK-44f5391a-625fb7dc-6c08
Max-Forwards: 70
Contact: <sip:0315400401@192.168.1.5>
Content-Type: application/SDP
Content-Length: 293

v=0
o=AudiocodesGW 689998298 689998179 IN IP4 192.168.1.38
s=Phone-Call
c=IN IP4 192.168.1.38
t=0 0
m=image 6002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 0: INVITE sip:0315400504@192.168.1.153 SIP/2.0 (44)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 1: From: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9 (75)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 2: To: "0315400504"<sip:0315400504@192.168.1.153>;tag=as7eb208ae (62)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 3: Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 (56)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 4: CSeq: 1 INVITE (14)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK-44f5391a-625fb7dc-6c08 (78)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 6: Max-Forwards: 70 (16)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 7: Contact: <sip:0315400401@192.168.1.5> (38)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 8: Content-Type: application/SDP (29)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 9: Content-Length: 293 (19)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4329 parse_request: Header 10:  (0)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: v=0 (3)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: o=AudiocodesGW 689998298 689998179 IN IP4 192.168.1.38 (55)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: s=Phone-Call (12)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: c=IN IP4 192.168.1.38 (22)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: t=0 0 (5)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: m=image 6002 udptl t38 (22)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=T38FaxVersion:0 (17)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=T38MaxBitRate:14400 (21)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=T38FaxMaxBuffer:1024 (22)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=T38FaxMaxDatagram:122 (23)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=T38FaxRateManagement:transferredTCF (37)
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4361 parse_request: Line: a=T38FaxUdpEC:t38UDPRedundancy (30)
--- (10 headers 12 lines)---
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4157 find_call: = Found Their Call ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153 Their Tag 5b16956-13c4-44f5390d-625f8523-32f9 Our tag: as7eb208ae
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:14007 handle_request: **** Received INVITE (5) - Command in SIP INVITE
Sending to 192.168.1.5 : 5060 (NAT)
Got T.38 offer in SDP in dialog 106aec5b00b8145d46db39c050c4d425@192.168.1.153
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4578 process_sdp: T38 state changed to 4 on channel SIP/ssw-007b4d30
Peer doesn't provide audio
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4644 process_sdp: Peer T.38 UDPTL is at port 192.168.1.38:6002
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4761 process_sdp: FaxVersion: 0
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4736 process_sdp: T38MaxBitRate: 14400
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4731 process_sdp: MaxBufferSize:1024
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4770 process_sdp: FaxMaxDatagram: 122
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4798 process_sdp: RateMangement: transferredTCF
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4807 process_sdp: UDP EC: t38UDPRedundancy
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:4827 process_sdp: Our T38 capability = (3872), peer T38 capability (16160), joint T38 capability (3872)
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Aug 30 09:03:26] NOTICE[26104]: chan_sip.c:4862 process_sdp: No compatible codecs, not accepting this offer!
Transmitting (NAT) to 192.168.1.5:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-44f5391a-625fb7dc-6c08;received=192.168.1.5;rport=5060
From: <sip:0315400401@192.168.1.5>;tag=5b16956-13c4-44f5390d-625f8523-32f9
To: "0315400504"<sip:0315400504@192.168.1.153>;tag=as7eb208ae
Call-ID: 106aec5b00b8145d46db39c050c4d425@192.168.1.153
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0315400504@192.168.1.153>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


---
[Aug 30 09:03:26] DEBUG[26104]: chan_sip.c:14218 sipsock_read: SIP message could not be handled, bad request: 106aec5b00b8145d46db39c050c4d425@192.168.1.153