[Mar 13 12:47:39] VERBOSE[10605] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.27:5060: INVITE sip:Anonymous@192.168.1.27:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.143:5060;branch=z9hG4bK023ccba2 Route: Max-Forwards: 70 From: ;tag=as19f8baf6 To: ;tag=gateway34854647rdb2214 Contact: Call-ID: 10597692-1-2930411748@192.168.1.22 CSeq: 104 INVITE User-Agent: Asterisk PBX 11.2.1 Session-Expires: 600;refresher=uas Min-SE: 180 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 261 v=0 o=root 87046065 87046069 IN IP4 172.16.1.143 s=Asterisk PBX 11.2.1 c=IN IP4 172.16.1.143 t=0 0 m=image 4235 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC --- [Mar 13 12:47:39] DEBUG[10605] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.1.27:5060 [Mar 13 12:47:39] WARNING[28838][C-0000bcde] res_fax.c: channel 'SIP/gateway-ny-0000a72f' timed-out during the T.38 negotiation. ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ [Mar 13 12:47:39] DEBUG[28838][C-0000bcde] res_rtp_asterisk.c: Difference is 40016, ms is 5022 [Mar 13 12:47:42] VERBOSE[10605] chan_sip.c: <--- SIP read from UDP:192.168.1.27:5060 ---> SIP/2.0 491 Server Internal Error - re-Invit Via: SIP/2.0/UDP 172.16.1.143:5060;branch=z9hG4bK023ccba2 To: ;tag=gateway34854647rdb2214 From: ;tag=as19f8baf6 Call-ID: 10597692-1-2930411748@192.168.1.22 CSeq: 104 INVITE Content-Length: 0 <-------------> [Mar 13 12:47:42] VERBOSE[10605] chan_sip.c: --- (7 headers 0 lines) --- [Mar 13 12:47:42] DEBUG[10605][C-0000bcde] logger.c: CALL_ID [C-0000bcde] bound to thread. [Mar 13 12:47:42] DEBUG[10605][C-0000bcde] chan_sip.c: Acked pending invite 104 [Mar 13 12:47:42] DEBUG[10605][C-0000bcde] chan_sip.c: Stopping retransmission on '10597692-1-2930411748@192.168.1.22' of Request 104: Match Found [Mar 13 12:47:42] VERBOSE[10605][C-0000bcde] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 13 12:47:42] VERBOSE[10605][C-0000bcde] chan_sip.c: set_destination: set destination to 192.168.1.27:5060 [Mar 13 12:47:42] VERBOSE[10605][C-0000bcde] chan_sip.c: Transmitting (no NAT) to 192.168.1.27:5060: ACK sip:Anonymous@192.168.1.27:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.143:5060;branch=z9hG4bK023ccba2 Route: Max-Forwards: 70 From: ;tag=as19f8baf6 To: ;tag=gateway34854647rdb2214 Contact: Call-ID: 10597692-1-2930411748@192.168.1.22 CSeq: 104 ACK User-Agent: Asterisk PBX 11.2.1 Content-Length: 0 --- [Mar 13 12:47:42] DEBUG[10605][C-0000bcde] chan_sip.c: Trying to put 'ACK sip:Ano' onto UDP socket destined for 192.168.1.27:5060 [Mar 13 12:47:42] WARNING[10605][C-0000bcde] chan_sip.c: just did sched_add waitid(228880) for sip_reinvite_retry for dialog 10597692-1-29304 11748@192.168.1.22 in handle_response_invite [Mar 13 12:47:42] DEBUG[10605][C-0000bcde] chan_sip.c: Reinvite race. Waiting 441 secs before retry [Mar 13 12:47:42] DEBUG[10605][C-0000bcde] logger.c: Call_ID [C-0000bcde] being removed from thread. [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] logger.c: CALL_ID [C-0000bcde] bound to thread. [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '10597692-1-2930411748@192.168.1.22' Request 105: Found [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] logger.c: Call_ID [C-0000bcde] being removed from thread. [Mar 13 12:47:43] DEBUG[10605] chan_sip.c: Sending pending reinvite on '10597692-1-2930411748@192.168.1.22' [Mar 13 12:47:43] VERBOSE[10605] chan_sip.c: set_destination: Parsing for add ress/port to send to [Mar 13 12:47:43] VERBOSE[10605] chan_sip.c: set_destination: set destination to 192.168.1.27:5060 [Mar 13 12:47:43] DEBUG[10605] chan_sip.c: T.38 UDPTL is at 172.16.1.143 port 4235 [Mar 13 12:47:43] DEBUG[10605] chan_sip.c: Done building SDP. Settling with this capability: (nothing) [Mar 13 12:47:43] DEBUG[10605] chan_sip.c: Initializing already initialized SIP dialog 10597692-1-2930411748@192.168.1.22 (presumably reinvi te) [Mar 13 12:47:43] VERBOSE[10605] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.27:5060: INVITE sip:Anonymous@192.168.1.27:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.143:5060;branch=z9hG4bK4ea2938a Route: Max-Forwards: 70 From: ;tag=as19f8baf6 To: ;tag=gateway34854647rdb2214 Contact: Call-ID: 10597692-1-2930411748@192.168.1.22 CSeq: 105 INVITE User-Agent: Asterisk PBX 11.2.1 Session-Expires: 600;refresher=uas Min-SE: 180 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 261 v=0 o=root 87046065 87046069 IN IP4 172.16.1.143 s=Asterisk PBX 11.2.1 c=IN IP4 172.16.1.143 t=0 0 m=image 4235 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:849 a=T38FaxUdpEC:t38UDPFEC --- [Mar 13 12:47:43] DEBUG[10605] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.1.27:5060 [Mar 13 12:47:43] VERBOSE[10605] chan_sip.c: <--- SIP read from UDP:192.168.1.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.1.143:5060;branch=z9hG4bK4ea2938a To: ;tag=gateway34854647rdb2214 From: ;tag=as19f8baf6 Call-ID: 10597692-1-2930411748@192.168.1.22 CSeq: 105 INVITE Content-Length: 0 <-------------> [Mar 13 12:47:43] VERBOSE[10605] chan_sip.c: --- (7 headers 0 lines) --- [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] logger.c: CALL_ID [C-0000bcde] bound to thread. [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '10597692-1-2930411748@192.168.1.22' Request 105: Found [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] logger.c: Call_ID [C-0000bcde] being removed from thread. [Mar 13 12:47:43] VERBOSE[10605] chan_sip.c: <--- SIP read from UDP:192.168.1.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.1.143:5060;branch=z9hG4bK4ea2938a Record-Route: To: ;tag=gateway34854647rdb2214 From: ;tag=as19f8baf6 Call-ID: 10597692-1-2930411748@192.168.1.22 CSeq: 105 INVITE Require: timer Session-Expires: 600;refresher=uas Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=Gateway-SBC 188 5 IN IP4 192.168.1.27 s=Session Controller c=IN IP4 192.168.92.22 t=0 0 m=image 60230 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Mar 13 12:47:43] VERBOSE[10605][C-0000bcde] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] chan_sip.c: Peer T.38 UDPTL is at port 192.168.92.22:60230 [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] chan_sip.c: T38 state changed to 3 on channel SIP/gateway-ny-0000a72f [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] chan_sip.c: Have T.38 but no audio, accepting offer anyway [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] chan_sip.c: Updating call counter for incoming call [Mar 13 12:47:43] VERBOSE[10605][C-0000bcde] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 13 12:47:43] VERBOSE[10605][C-0000bcde] chan_sip.c: set_destination: set destination to 192.168.1.27:5060 [Mar 13 12:47:43] VERBOSE[10605][C-0000bcde] chan_sip.c: Transmitting (no NAT) to 192.168.1.27:5060: ACK sip:Anonymous@192.168.1.27:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.143:5060;branch=z9hG4bK1480ecd0 Route: Max-Forwards: 70 From: ;tag=as19f8baf6 To: ;tag=gateway34854647rdb2214 Contact: Call-ID: 10597692-1-2930411748@192.168.1.22 CSeq: 105 ACK User-Agent: Asterisk PBX 11.2.1 Content-Length: 0 --- [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] chan_sip.c: Trying to put 'ACK sip:Ano' onto UDP socket destined for 192.168.1.27:5060 [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] logger.c: Call_ID [C-0000bcde] being removed from thread. [Mar 13 12:47:43] DEBUG[28838][C-0000bcde] pbx.c: Function result is 'Anonymous' [Mar 13 12:47:43] VERBOSE[28838][C-0000bcde] res_fax.c: -- Channel 'SIP/gateway-ny-0000a72f' switched to T.38 FAX session '1115'. [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] chan_sip.c: Updating call counter for incoming call [Mar 13 12:47:43] VERBOSE[10605][C-0000bcde] chan_sip.c: set_destination: Parsing for address/port to send to [Mar 13 12:47:43] VERBOSE[10605][C-0000bcde] chan_sip.c: set_destination: set destination to 192.168.1.27:5060 [Mar 13 12:47:43] VERBOSE[10605][C-0000bcde] chan_sip.c: Transmitting (no NAT) to 192.168.1.27:5060: ACK sip:Anonymous@192.168.1.27:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.143:5060;branch=z9hG4bK1480ecd0 Route: Max-Forwards: 70 From: ;tag=as19f8baf6 To: ;tag=gateway34854647rdb2214 Contact: Call-ID: 10597692-1-2930411748@192.168.1.22 CSeq: 105 ACK User-Agent: Asterisk PBX 11.2.1 Content-Length: 0 --- [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] chan_sip.c: Trying to put 'ACK sip:Ano' onto UDP socket destined for 192.168.1.27:5060 [Mar 13 12:47:43] DEBUG[10605][C-0000bcde] logger.c: Call_ID [C-0000bcde] being removed from thread. [Mar 13 12:47:43] DEBUG[28838][C-0000bcde] pbx.c: Function result is 'Anonymous' [Mar 13 12:47:43] VERBOSE[28838][C-0000bcde] res_fax.c: -- Channel 'SIP/gateway-ny-0000a72f' switched to T.38 FAX session '1115'. [Mar 13 12:47:43] DEBUG[10626] manager.c: Examining event: Event: FAXStatus Privilege: call,all Operation: receive Status: T.38 Negotiated Channel: SIP/gateway-ny-0000a72f Context: ivr-nofax Exten: s CallerID: Anonymous LocalStationID: unknown