=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.07.13 19:17:39 =~=~=~=~=~=~=~=~=~=~=~= asterisk -c -n Asterisk 1.2.9, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= [ Booting...Jul 13 19:29:53 NOTICE[17655]: cdr.c:1191 do_reload: CDR simple logging enabled. .................Jul 13 19:29:53 ERROR[17655]: netsock.c:140 ast_netsock_bindaddr: Unable to bind to 212.37.49.98 port 4569: Cannot assign requested address Jul 13 19:29:53 WARNING[17655]: chan_iax2.c:8698 set_config: Unable apply binding to '212.37.49.98' at line 17 Jul 13 19:29:53 WARNING[17655]: chan_iax2.c:8305 build_peer: Set peer->pokefreqnotok to 10000 Jul 13 19:29:53 WARNING[17655]: chan_iax2.c:8305 build_peer: Set peer->pokefreqnotok to 10000 .... -- == Setting default context to default ........................................................................................................ ] Asterisk Ready. ]1;Asterisk]2;Asterisk Console on 'linux' (pid 17655)*CLI> set debug 4 Core debug was 0 and is now 4 *CLI> set verJul 13 19:29:57 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #20)) bJul 13 19:29:57 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #25)) ose 4 Verbosity was 0 and is now 4 *CLI> sip debug SIP Debugging enabled *CLI> Jul 13 19:30:01 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #20 (4) REGISTER - 2 Jul 13 19:30:01 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #20)) Retransmitting #4 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK71f4cc80;rport From: ;tag=as59641e74 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- *CLI> Jul 13 19:30:01 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #25 (4) REGISTER - 2 Jul 13 19:30:01 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #25)) Retransmitting #4 (no NAT) to 217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK6e07887f;rport From: ;tag=as5e0fc633 To: Call-ID: 718f233b7a1f8f08321df0d82be766d0@127.0.0.2 CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:04 DEBUG[17659]: chan_iax2.c:9347 iax2_devicestate: Checking device state for device wf Jul 13 19:30:04 DEBUG[17659]: chan_iax2.c:9355 iax2_devicestate: iax2_devicestate(wf): Found peer. What's device state of wf? addr=16820416, defaddr=0 maxms=2000, lastms=28 Jul 13 19:30:04 DEBUG[17659]: devicestate.c:187 do_state_change: Changing state for IAX2/wf - state 1 (Not in use) Jul 13 19:30:04 DEBUG[17694]: app_queue.c:523 changethread: Device 'IAX2/wf' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jul 13 19:30:04 DEBUG[17659]: chan_iax2.c:9347 iax2_devicestate: Checking device state for device wf Jul 13 19:30:04 DEBUG[17659]: chan_iax2.c:9355 iax2_devicestate: iax2_devicestate(wf): Found peer. What's device state of wf? addr=16820416, defaddr=0 maxms=2000, lastms=28 Jul 13 19:30:04 DEBUG[17659]: devicestate.c:187 do_state_change: Changing state for IAX2/wf - state 1 (Not in use) Jul 13 19:30:04 DEBUG[17695]: app_queue.c:523 changethread: Device 'IAX2/wf' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jul 13 19:30:05 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #20 (5) REGISTER - 2 Jul 13 19:30:05 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #20)) Retransmitting #5 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK71f4cc80;rport From: ;tag=as59641e74 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:05 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #25 (5) REGISTER - 2 Jul 13 19:30:05 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #25)) Retransmitting #5 (no NAT) to 217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK6e07887f;rport From: ;tag=as5e0fc633 To: Call-ID: 718f233b7a1f8f08321df0d82be766d0@127.0.0.2 CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #20 (6) REGISTER - 2 Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #20)) Retransmitting #6 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK71f4cc80;rport From: ;tag=as59641e74 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #25 (6) REGISTER - 2 Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #25)) Retransmitting #6 (no NAT) to 217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK6e07887f;rport From: ;tag=as5e0fc633 To: Call-ID: 718f233b7a1f8f08321df0d82be766d0@127.0.0.2 CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 217.10.79.9:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK6e07887f;rport=21746;received=212.37.49.98 From: ;tag=as5e0fc633 To: ;tag=b11cb9bb270104b49a99a995b8c68544.d7d3 Call-ID: 718f233b7a1f8f08321df0d82be766d0@127.0.0.2 CSeq: 102 REGISTER WWW-Authenticate: Digest realm="sipgate.de", nonce="44b68175bd8f901c36a306c9f6099393160e1815" Server: sipgate ser Content-Length: 0 Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: SIP/2.0 401 Unauthorized (24) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK6e07887f;rport=21746;received=212.37.49.98 (92) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=as5e0fc633 (45) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=b11cb9bb270104b49a99a995b8c68544.d7d3 (70) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 718f233b7a1f8f08321df0d82be766d0@127.0.0.2 (51) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 102 REGISTER (18) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: WWW-Authenticate: Digest realm="sipgate.de", nonce="44b68175bd8f901c36a306c9f6099393160e1815" (93) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Server: sipgate ser (19) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Content-Length: 0 (17) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25 Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '718f233b7a1f8f08321df0d82be766d0@127.0.0.2' of Request 102: Match Found Responding to challenge, registration to domain/host name sipgate.de Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: REGISTER sip:sipgate.de SIP/2.0 (31) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK503bcaad;rport (64) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=as336a6761 (45) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: (28) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 718f233b7a1f8f08321df0d82be766d0@127.0.0.2 (51) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 103 REGISTER (18) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: Asterisk PBX (24) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Max-Forwards: 70 (16) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Authorization: Digest username="9879010", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="44b68175bd8f901c36a306c9f6099393160e1815", response="f0ac9bb76248998121c65869970be478", opaque="" (203) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Expires: 1200 (13) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: Contact: (36) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: Event: registration (19) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 12: Content-Length: 0 (17) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 13: (0) REGISTER 13 headers, 0 lines REGISTER attempt 2 to 9879010@sipgate.de Reliably Transmitting (no NAT) to 217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK503bcaad;rport From: ;tag=as336a6761 To: Call-ID: 718f233b7a1f8f08321df0d82be766d0@127.0.0.2 CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="9879010", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="44b68175bd8f901c36a306c9f6099393160e1815", response="f0ac9bb76248998121c65869970be478", opaque="" Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #28 <-- SIP read from 217.10.79.9:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK503bcaad;rport=21746;received=212.37.49.98 From: ;tag=as336a6761 To: ;tag=b11cb9bb270104b49a99a995b8c68544.144c Call-ID: 718f233b7a1f8f08321df0d82be766d0@127.0.0.2 CSeq: 103 REGISTER Contact: ;expires=1200 Server: sipgate ser Content-Length: 0 Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: SIP/2.0 200 OK (14) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK503bcaad;rport=21746;received=212.37.49.98 (92) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=as336a6761 (45) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=b11cb9bb270104b49a99a995b8c68544.144c (70) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 718f233b7a1f8f08321df0d82be766d0@127.0.0.2 (51) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 103 REGISTER (18) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: Contact: ;expires=1200 (54) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Server: sipgate ser (19) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Content-Length: 0 (17) Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #28 Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '718f233b7a1f8f08321df0d82be766d0@127.0.0.2' of Request 103: Match Found Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:9778 handle_response_register: Registration successful Jul 13 19:30:09 DEBUG[17674]: chan_sip.c:9780 handle_response_register: Cancelling timeout 24 Scheduling destruction of call '718f233b7a1f8f08321df0d82be766d0@127.0.0.2' in 32000 ms Jul 13 19:30:09 NOTICE[17674]: chan_sip.c:9830 handle_response_register: Outbound Registration: Expiry for sipgate.de is 1200 sec (Scheduling reregistration in 1185 s) <-- SIP read from 82.165.39.16:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 192.168.0.41:5060: INVITE sip:100@192.168.0.160:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T3F87EB8D Session-Expires: 90;refresher=uas From: "200" ;tag=000FC9014CE3_T274886762 To: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103792 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Supported: timer,replaces User-Agent: ALL7950 02.09.18 Content-Type: application/sdp Content-Length: 293 v=0 o=200 275878517 275878517 IN IP4 192.168.0.41 s=ALL7950 02.09.18 c=IN IP4 192.168.0.41 t=0 0 m=audio 8000 RTP/AVP 8 18 4 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: INVITE sip:100@192.168.0.160:5060 SIP/2.0 (41) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T3F87EB8D (72) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: Session-Expires: 90;refresher=uas (33) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: From: "200" ;tag=000FC9014CE3_T274886762 (68) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: To: (32) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 (54) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: CSeq: 468103792 INVITE (22) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Contact: (36) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO (74) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: Supported: timer,replaces (25) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: User-Agent: ALL7950 02.09.18 (28) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 12: Content-Type: application/sdp (29) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 13: Content-Length: 293 (19) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 14: (0) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: v=0 (3) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: o=200 275878517 275878517 IN IP4 192.168.0.41 (45) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: s=ALL7950 02.09.18 (18) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: c=IN IP4 192.168.0.41 (21) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: t=0 0 (5) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: m=audio 8000 RTP/AVP 8 18 4 101 (31) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:8 PCMA/8000/1 (22) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:18 G729/8000/1 (23) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:18 annexb=no (19) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:4 G723/8000/1 (22) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:101 0-15 (15) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=sendrecv (10) --- (14 headers 13 lines)--- Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 - INVITE (With RTP) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received INVITE (5) - Command in SIP INVITE Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1004 parse_sip_options: Begin: parsing SIP "Supported: timer,replaces" Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1016 parse_sip_options: Found SIP option: -timer- Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1022 parse_sip_options: Matched SIP option: timer Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1016 parse_sip_options: Found SIP option: -replaces- Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1022 parse_sip_options: Matched SIP option: replaces Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1033 parse_sip_options: * SIP extension value: 5 for call CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 Using INVITE request as basis request - CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 Sending to 192.168.0.41 : 5060 (non-NAT) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:7155 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 192.168.0.41:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T3F87EB8D;received=192.168.0.41 From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as1ed01d30 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103792 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="509a04fe" Content-Length: 0 --- Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #31 Scheduling destruction of call 'CALL_ID9_000FC9014CE3_T325875288@192.168.0.41' in 15000 ms Found user '200' <-- SIP read from 192.168.0.41:5060: ACK sip:100@192.168.0.160:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T3F87EB8D From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as1ed01d30 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103792 ACK User-Agent: ALL7950 02.09.18 Contact: Max-Forwards: 70 Content-Length: 0 Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: ACK sip:100@192.168.0.160:5060 SIP/2.0 (38) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T3F87EB8D (72) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=000FC9014CE3_T274886762 (68) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=as1ed01d30 (47) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 (54) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 468103792 ACK (19) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: ALL7950 02.09.18 (28) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Contact: (36) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #31 Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on 'CALL_ID9_000FC9014CE3_T325875288@192.168.0.41' of Response 468103792: Match Found <-- SIP read from 192.168.0.41:5060: INVITE sip:100@192.168.0.160:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T10626232 Session-Expires: 90;refresher=uas From: "200" ;tag=000FC9014CE3_T274886762 To: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103793 INVITE Proxy-Authorization: Digest username="200", realm="asterisk", nonce="509a04fe", opaque="", uri="sip:100@192.168.0.160:5060", response="bca2fcc8362e077d6cc90ecc9f514285" Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Supported: timer,replaces User-Agent: ALL7950 02.09.18 Content-Type: application/sdp Content-Length: 293 v=0 o=200 275878517 275878517 IN IP4 192.168.0.41 s=ALL7950 02.09.18 c=IN IP4 192.168.0.41 t=0 0 m=audio 8000 RTP/AVP 8 18 4 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: INVITE sip:100@192.168.0.160:5060 SIP/2.0 (41) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T10626232 (72) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: Session-Expires: 90;refresher=uas (33) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: From: "200" ;tag=000FC9014CE3_T274886762 (68) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: To: (32) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 (54) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: CSeq: 468103793 INVITE (22) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Proxy-Authorization: Digest username="200", realm="asterisk", nonce="509a04fe", opaque="", uri="sip:100@192.168.0.160:5060", response="bca2fcc8362e077d6cc90ecc9f514285" (168) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Contact: (36) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Max-Forwards: 70 (16) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO (74) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: Supported: timer,replaces (25) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 12: User-Agent: ALL7950 02.09.18 (28) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 13: Content-Type: application/sdp (29) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 14: Content-Length: 293 (19) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 15: (0) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: v=0 (3) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: o=200 275878517 275878517 IN IP4 192.168.0.41 (45) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: s=ALL7950 02.09.18 (18) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: c=IN IP4 192.168.0.41 (21) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: t=0 0 (5) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: m=audio 8000 RTP/AVP 8 18 4 101 (31) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:8 PCMA/8000/1 (22) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:18 G729/8000/1 (23) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:18 annexb=no (19) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:4 G723/8000/1 (22) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:101 0-15 (15) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=sendrecv (10) --- (15 headers 13 lines)--- Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 Sending to 192.168.0.41 : 5060 (non-NAT) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:7155 check_user_full: Setting NAT on RTP to 0 Found user '200' Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.41:8000 Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 192.168.0.41:8000 Found description format PCMA Found description format G729 Found description format G723 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x109 (g723|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:10497 handle_request_invite: Checking SIP call limits for device 200 Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:2209 update_call_counter: Updating call counter for incoming call Looking for 100 in Softphone (domain 192.168.0.160) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:6137 build_route: build_route: Contact hop: list_route: hop: Transmitting (no NAT) to 192.168.0.41:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T10626232;received=192.168.0.41 From: "200" ;tag=000FC9014CE3_T274886762 To: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103793 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 13 19:30:10 DEBUG[17659]: chan_sip.c:11668 sip_devicestate: Checking device state for peer 200 Jul 13 19:30:10 DEBUG[17659]: devicestate.c:187 do_state_change: Changing state for SIP/200 - state 2 (In use) Jul 13 19:30:10 DEBUG[17696]: pbx.c:1677 pbx_extension_helper: Launching 'NoOp' -- Executing NoOp("SIP/200-6a41", "") in new stack Jul 13 19:30:10 DEBUG[17696]: pbx.c:1677 pbx_extension_helper: Launching 'Dial' -- Executing Dial("SIP/200-6a41", "SIP/100") in new stack Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:1874 create_addr_from_peer: Setting NAT on RTP to 0 Jul 13 19:30:10 DEBUG[17696]: channel.c:2823 ast_channel_inherit_variables: Not copying variable STACK-Softphone-100-2. Jul 13 19:30:10 DEBUG[17696]: channel.c:2823 ast_channel_inherit_variables: Not copying variable STACK-Softphone-100-1. Jul 13 19:30:10 DEBUG[17696]: channel.c:2823 ast_channel_inherit_variables: Not copying variable SIPCALLID. Jul 13 19:30:10 DEBUG[17696]: channel.c:2823 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. Jul 13 19:30:10 DEBUG[17696]: channel.c:2823 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. Jul 13 19:30:10 DEBUG[17696]: channel.c:2823 ast_channel_inherit_variables: Not copying variable SIPURI. Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:2068 sip_call: Outgoing Call for 100 Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:2209 update_call_counter: Updating call counter for outgoing call We're at 192.168.0.160 port 18438 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 0: INVITE sip:100@192.168.0.50:5060 SIP/2.0 (40) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK58bc19c0;rport (64) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=as4dd76d35 (50) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 3: To: (31) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 4: Contact: (32) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 5: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 7: User-Agent: Asterisk PBX (24) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 9: Date: Thu, 13 Jul 2006 17:30:10 GMT (35) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 11: Content-Type: application/sdp (29) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 12: Content-Length: 296 (19) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3363 parse_request: Header 13: (0) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: v=0 (3) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: o=root 17655 17655 IN IP4 192.168.0.160 (39) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: s=session (9) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: c=IN IP4 192.168.0.160 (22) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: t=0 0 (5) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: m=audio 18438 RTP/AVP 8 0 111 3 101 (35) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: a=rtpmap:111 G726-32/8000 (25) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: a=fmtp:101 0-16 (15) Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:3395 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 13 lines Reliably Transmitting (no NAT) to 192.168.0.50:5060: INVITE sip:100@192.168.0.50:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK58bc19c0;rport From: "200" ;tag=as4dd76d35 To: Contact: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 13 Jul 2006 17:30:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 296 v=0 o=root 17655 17655 IN IP4 192.168.0.160 s=session c=IN IP4 192.168.0.160 t=0 0 m=audio 18438 RTP/AVP 8 0 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 13 19:30:10 DEBUG[17696]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #33 -- Called 100 Jul 13 19:30:10 DEBUG[17696]: channel.c:2350 set_format: Set channel SIP/100-3e19 to read format slin Jul 13 19:30:10 DEBUG[17696]: channel.c:2350 set_format: Set channel SIP/200-6a41 to write format slin Jul 13 19:30:10 DEBUG[17696]: channel.c:2350 set_format: Set channel SIP/200-6a41 to read format slin Jul 13 19:30:10 DEBUG[17696]: channel.c:2350 set_format: Set channel SIP/100-3e19 to write format slin Jul 13 19:30:10 DEBUG[17697]: app_queue.c:523 changethread: Device 'SIP/200' changed to state '2' (In use) but we don't care because they're not a member of any queue. <-- SIP read from 192.168.0.50:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK58bc19c0;rport From: "200" ;tag=as4dd76d35 To: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 Contact: CSeq: 102 INVITE User-Agent: ALL7950 02.09.26 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Content-Length: 0 Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: SIP/2.0 100 Trying (18) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK58bc19c0;rport (64) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=as4dd76d35 (50) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: (31) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: Contact: (36) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: User-Agent: ALL7950 02.09.26 (28) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO (74) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1445 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #33 - INVITE (got response) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1454 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160' Request 102: Found Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:9584 handle_response_invite: SIP response 100 to standard invite <-- SIP read from 192.168.0.50:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK58bc19c0;rport From: "200" ;tag=as4dd76d35 To: ;tag=000FC9015027_T2012322891 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 Contact: CSeq: 102 INVITE User-Agent: ALL7950 02.09.26 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Content-Length: 0 Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: SIP/2.0 180 Ringing (19) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK58bc19c0;rport (64) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=as4dd76d35 (50) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=000FC9015027_T2012322891 (60) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: Contact: (36) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: User-Agent: ALL7950 02.09.26 (28) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO (74) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:1454 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160' Request 102: Found Jul 13 19:30:10 DEBUG[17674]: chan_sip.c:9584 handle_response_invite: SIP response 180 to standard invite Jul 13 19:30:10 DEBUG[17659]: chan_sip.c:11668 sip_devicestate: Checking device state for peer 100 Jul 13 19:30:10 DEBUG[17659]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/100-3e19' -- SIP/100-3e19 is ringing Transmitting (no NAT) to 192.168.0.41:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T10626232;received=192.168.0.41 From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as018c72c1 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103793 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 13 19:30:10 DEBUG[17659]: devicestate.c:187 do_state_change: Changing state for SIP/100 - state 6 (Ringing) Jul 13 19:30:10 DEBUG[17698]: app_queue.c:523 changethread: Device 'SIP/100' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. <-- SIP read from 217.10.79.9:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 192.168.0.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK58bc19c0;rport From: "200" ;tag=as4dd76d35 To: ;tag=000FC9015027_T2012322891 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 Contact: CSeq: 102 INVITE Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Supported: timer,replaces Content-Type: application/sdp Content-Length: 218 v=0 o=100 290009649 290009649 IN IP4 192.168.0.50 s=ALL7950 02.09.26 c=IN IP4 192.168.0.50 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: SIP/2.0 200 OK (14) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK58bc19c0;rport (64) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=as4dd76d35 (50) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=000FC9015027_T2012322891 (60) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: Contact: (36) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO (74) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Supported: timer,replaces (25) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Type: application/sdp (29) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: Content-Length: 218 (19) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: (0) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: v=0 (3) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: o=100 290009649 290009649 IN IP4 192.168.0.50 (45) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: s=ALL7950 02.09.26 (18) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: c=IN IP4 192.168.0.50 (21) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: t=0 0 (5) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: m=audio 8000 RTP/AVP 8 101 (26) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:8 PCMA/8000/1 (22) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:101 0-15 (15) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=sendrecv (10) --- (11 headers 10 lines)--- Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160' of Request 102: Match Found Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:9584 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.50:8000 Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 192.168.0.50:8000 Found description format PCMA Found description format telephone-event Capabilities: us - 0x1e (gsm|ulaw|alaw|g726), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3701 process_sdp: Oooh, we need to change our formats since our peer supports only 0x8 (alaw) and not 0x4 (ulaw) Jul 13 19:30:13 DEBUG[17674]: channel.c:2350 set_format: Set channel SIP/100-3e19 to read format slin Jul 13 19:30:13 DEBUG[17674]: channel.c:2350 set_format: Set channel SIP/100-3e19 to write format slin Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:6137 build_route: build_route: Contact hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.50, port 5060 Transmitting (no NAT) to 192.168.0.50:5060: ACK sip:100@192.168.0.50:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK0e71317c;rport From: "200" ;tag=as4dd76d35 To: ;tag=000FC9015027_T2012322891 Contact: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/100-3e19 answered SIP/200-6a41 Jul 13 19:30:13 DEBUG[17696]: channel.c:2350 set_format: Set channel SIP/200-6a41 to read format alaw Jul 13 19:30:13 DEBUG[17696]: channel.c:2350 set_format: Set channel SIP/100-3e19 to write format alaw Jul 13 19:30:13 DEBUG[17696]: channel.c:2350 set_format: Set channel SIP/100-3e19 to read format alaw Jul 13 19:30:13 DEBUG[17696]: channel.c:2350 set_format: Set channel SIP/200-6a41 to write format alaw Jul 13 19:30:13 DEBUG[17696]: chan_sip.c:2540 sip_answer: sip_answer(SIP/200-6a41) We're at 192.168.0.160 port 13652 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T10626232;received=192.168.0.41 From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as018c72c1 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103793 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 17655 17655 IN IP4 192.168.0.160 s=session c=IN IP4 192.168.0.160 t=0 0 m=audio 13652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 13 19:30:13 DEBUG[17696]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #35 -- Attempting native bridge of SIP/200-6a41 and SIP/100-3e19 Jul 13 19:30:13 DEBUG[17659]: chan_sip.c:11668 sip_devicestate: Checking device state for peer 100 Jul 13 19:30:13 DEBUG[17659]: devicestate.c:187 do_state_change: Changing state for SIP/100 - state 2 (In use) Jul 13 19:30:13 DEBUG[17659]: chan_sip.c:11668 sip_devicestate: Checking device state for peer 200 Jul 13 19:30:13 DEBUG[17659]: devicestate.c:187 do_state_change: Changing state for SIP/200 - state 2 (In use) Jul 13 19:30:13 DEBUG[17699]: app_queue.c:523 changethread: Device 'SIP/100' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 13 19:30:13 DEBUG[17700]: app_queue.c:523 changethread: Device 'SIP/200' changed to state '2' (In use) but we don't care because they're not a member of any queue. <-- SIP read from 192.168.0.41:5060: ACK sip:100@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T124B9659 From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as018c72c1 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103793 ACK User-Agent: ALL7950 02.09.18 Contact: Max-Forwards: 70 Content-Length: 0 Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: ACK sip:100@192.168.0.160 SIP/2.0 (33) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T124B9659 (72) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=000FC9014CE3_T274886762 (68) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=as018c72c1 (47) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 (54) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 468103793 ACK (19) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: ALL7950 02.09.18 (28) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Contact: (36) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #35 Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on 'CALL_ID9_000FC9014CE3_T325875288@192.168.0.41' of Response 468103793: Match Found Jul 13 19:30:13 DEBUG[17696]: rtp.c:1352 ast_rtp_write: Ooh, format changed from unknown to alaw Jul 13 19:30:13 DEBUG[17696]: rtp.c:1352 ast_rtp_write: Ooh, format changed from unknown to alaw Jul 13 19:30:13 NOTICE[17674]: chan_sip.c:5372 sip_reg_timeout: -- Registration for 'sip22660@sip.sipport.de' timed out, trying again (Attempt #1) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #20 Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '2859f57d4fd8882647603bc55c4536b5@127.0.0.2' of Request 102: Match Found Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 - REGISTER (No RTP) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:5507 transmit_register: Scheduled a registration timeout for sip.sipport.de id #36 Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: REGISTER sip:sip.sipport.de SIP/2.0 (35) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK51707cbc;rport (64) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=as36851129 (50) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: (33) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 (51) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 103 REGISTER (18) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: Asterisk PBX (24) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Max-Forwards: 70 (16) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Expires: 1200 (13) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Contact: (36) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: Event: registration (19) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: Content-Length: 0 (17) Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 12: (0) REGISTER 12 headers, 0 lines REGISTER attempt 2 to sip22660@sip.sipport.de Reliably Transmitting (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK51707cbc;rport From: ;tag=as36851129 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:13 DEBUG[17674]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #37 Destroying call '2859f57d4fd8882647603bc55c4536b5@127.0.0.2' Jul 13 19:30:14 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (1) REGISTER - 2 Jul 13 19:30:14 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #37)) Retransmitting #1 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK51707cbc;rport From: ;tag=as36851129 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:15 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (2) REGISTER - 2 Jul 13 19:30:15 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #37)) Retransmitting #2 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK51707cbc;rport From: ;tag=as36851129 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:17 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (3) REGISTER - 2 Jul 13 19:30:17 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #37)) Retransmitting #3 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK51707cbc;rport From: ;tag=as36851129 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 192.168.0.41:5060: INVITE sip:100@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T128CCA23 From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as018c72c1 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103794 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Supported: timer,replaces User-Agent: ALL7950 02.09.18 Content-Type: application/sdp Content-Length: 288 v=0 o=200 275878517 275878518 IN IP4 192.168.0.41 s=ALL7950 02.09.18 c=IN IP4 0.0.0.0 t=0 0 m=audio 8000 RTP/AVP 8 18 4 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: INVITE sip:100@192.168.0.160 SIP/2.0 (36) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T128CCA23 (72) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=000FC9014CE3_T274886762 (68) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=as018c72c1 (47) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 (54) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 468103794 INVITE (22) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: Contact: (36) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Max-Forwards: 70 (16) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO (74) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Supported: timer,replaces (25) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: User-Agent: ALL7950 02.09.18 (28) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: Content-Type: application/sdp (29) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 12: Content-Length: 288 (19) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 13: (0) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: v=0 (3) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: o=200 275878517 275878518 IN IP4 192.168.0.41 (45) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: s=ALL7950 02.09.18 (18) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: c=IN IP4 0.0.0.0 (16) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: t=0 0 (5) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: m=audio 8000 RTP/AVP 8 18 4 101 (31) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:8 PCMA/8000/1 (22) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:18 G729/8000/1 (23) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:18 annexb=no (19) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:4 G723/8000/1 (22) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:101 0-15 (15) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=sendonly (10) --- (13 headers 13 lines)--- Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 Sending to 192.168.0.41 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:8000 Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 0.0.0.0:8000 Found description format PCMA Found description format G729 Found description format G723 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x109 (g723|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 13 19:30:18 DEBUG[17674]: channel.c:2350 set_format: Set channel SIP/100-3e19 to write format slin -- Started music on hold, class 'default', on channel 'SIP/100-3e19' Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:10551 handle_request_invite: Got a SIP re-invite for call CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 We're at 192.168.0.160 port 13652 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T128CCA23;received=192.168.0.41 From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as018c72c1 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103794 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 17655 17656 IN IP4 192.168.0.160 s=session c=IN IP4 192.168.0.160 t=0 0 m=audio 13652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #38 Jul 13 19:30:18 DEBUG[17696]: rtp.c:1260 ast_rtp_raw_write: Difference is 1296, ms is 182 <-- SIP read from 192.168.0.41:5060: ACK sip:100@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T0542DC52 From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as018c72c1 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103794 ACK User-Agent: ALL7950 02.09.18 Contact: Max-Forwards: 70 Content-Length: 0 Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: ACK sip:100@192.168.0.160 SIP/2.0 (33) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T0542DC52 (72) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=000FC9014CE3_T274886762 (68) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=as018c72c1 (47) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 (54) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 468103794 ACK (19) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: ALL7950 02.09.18 (28) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Contact: (36) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #38 Jul 13 19:30:18 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on 'CALL_ID9_000FC9014CE3_T325875288@192.168.0.41' of Response 468103794: Match Found Jul 13 19:30:21 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (4) REGISTER - 2 Jul 13 19:30:21 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #37)) Retransmitting #4 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK51707cbc;rport From: ;tag=as36851129 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:25 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (5) REGISTER - 2 Jul 13 19:30:25 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #37)) Retransmitting #5 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK51707cbc;rport From: ;tag=as36851129 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 82.165.39.16:5060: --- (0 headers 0 lines) Nat keepalive --- Jul 13 19:30:25 DEBUG[17674]: chan_sip.c:1323 __sip_autodestruct: Auto destroying call '00348f4100362818144f0ef658ed515a@127.0.0.2' Destroying call '00348f4100362818144f0ef658ed515a@127.0.0.2' <-- SIP read from 217.10.79.9:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 192.168.0.41:5060: INVITE sip:100@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T34A52D2C From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as018c72c1 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103795 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Supported: timer,replaces User-Agent: ALL7950 02.09.18 Content-Type: application/sdp Content-Length: 218 v=0 o=200 275878517 275878519 IN IP4 192.168.0.41 s=ALL7950 02.09.18 c=IN IP4 192.168.0.41 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: INVITE sip:100@192.168.0.160 SIP/2.0 (36) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T34A52D2C (72) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=000FC9014CE3_T274886762 (68) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=as018c72c1 (47) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 (54) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 468103795 INVITE (22) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: Contact: (36) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Max-Forwards: 70 (16) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO (74) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Supported: timer,replaces (25) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: User-Agent: ALL7950 02.09.18 (28) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: Content-Type: application/sdp (29) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 12: Content-Length: 218 (19) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 13: (0) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: v=0 (3) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: o=200 275878517 275878519 IN IP4 192.168.0.41 (45) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: s=ALL7950 02.09.18 (18) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: c=IN IP4 192.168.0.41 (21) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: t=0 0 (5) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: m=audio 8000 RTP/AVP 8 101 (26) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:8 PCMA/8000/1 (22) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:101 0-15 (15) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=sendrecv (10) --- (13 headers 10 lines)--- Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 Sending to 192.168.0.41 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.41:8000 Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 192.168.0.41:8000 Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 13 19:30:27 DEBUG[17674]: channel.c:2350 set_format: Set channel SIP/100-3e19 to write format alaw -- Stopped music on hold on SIP/100-3e19 Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:10551 handle_request_invite: Got a SIP re-invite for call CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 We're at 192.168.0.160 port 13652 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T34A52D2C;received=192.168.0.41 From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as018c72c1 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103795 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 17655 17657 IN IP4 192.168.0.160 s=session c=IN IP4 192.168.0.160 t=0 0 m=audio 13652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #39 <-- SIP read from 192.168.0.41:5060: ACK sip:100@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T74C5DCCD From: "200" ;tag=000FC9014CE3_T274886762 To: ;tag=as018c72c1 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 468103795 ACK User-Agent: ALL7950 02.09.18 Contact: Max-Forwards: 70 Content-Length: 0 Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: ACK sip:100@192.168.0.160 SIP/2.0 (33) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK_000FC9014CE3_T74C5DCCD (72) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: "200" ;tag=000FC9014CE3_T274886762 (68) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: ;tag=as018c72c1 (47) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 (54) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 468103795 ACK (19) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: ALL7950 02.09.18 (28) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Contact: (36) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #39 Jul 13 19:30:27 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on 'CALL_ID9_000FC9014CE3_T325875288@192.168.0.41' of Response 468103795: Match Found Jul 13 19:30:29 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (6) REGISTER - 2 Jul 13 19:30:29 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #37)) Retransmitting #6 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK51707cbc;rport From: ;tag=as36851129 To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:33 NOTICE[17674]: chan_sip.c:5372 sip_reg_timeout: -- Registration for 'sip22660@sip.sipport.de' timed out, trying again (Attempt #2) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #37 Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '2859f57d4fd8882647603bc55c4536b5@127.0.0.2' of Request 103: Match Found Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 - REGISTER (No RTP) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:5507 transmit_register: Scheduled a registration timeout for sip.sipport.de id #40 Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: REGISTER sip:sip.sipport.de SIP/2.0 (35) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK79cb4c6b;rport (64) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=as43326fea (50) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: (33) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 (51) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 104 REGISTER (18) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: Asterisk PBX (24) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Max-Forwards: 70 (16) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Expires: 1200 (13) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Contact: (36) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: Event: registration (19) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: Content-Length: 0 (17) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 12: (0) REGISTER 12 headers, 0 lines REGISTER attempt 3 to sip22660@sip.sipport.de Reliably Transmitting (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK79cb4c6b;rport From: ;tag=as43326fea To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #41 Destroying call '2859f57d4fd8882647603bc55c4536b5@127.0.0.2' <-- SIP read from 192.168.0.50:5060: INVITE sip:200@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T3CC17349 From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200156 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Supported: timer,replaces User-Agent: ALL7950 02.09.26 Content-Type: application/sdp Content-Length: 239 v=0 o=100 290009649 290009650 IN IP4 192.168.0.50 s=ALL7950 02.09.26 c=IN IP4 0.0.0.0 t=0 0 m=audio 8000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: INVITE sip:200@192.168.0.160 SIP/2.0 (36) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T3CC17349 (72) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=000FC9015027_T2012322891 (62) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: "200" ;tag=as4dd76d35 (48) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 1391200156 INVITE (23) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: Contact: (36) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Max-Forwards: 70 (16) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO (74) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Supported: timer,replaces (25) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: User-Agent: ALL7950 02.09.26 (28) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: Content-Type: application/sdp (29) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 12: Content-Length: 239 (19) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 13: (0) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: v=0 (3) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: o=100 290009649 290009650 IN IP4 192.168.0.50 (45) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: s=ALL7950 02.09.26 (18) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: c=IN IP4 0.0.0.0 (16) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: t=0 0 (5) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: m=audio 8000 RTP/AVP 8 0 101 (28) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:8 PCMA/8000/1 (22) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:0 PCMU/8000/1 (22) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:101 0-15 (15) Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=sendonly (10) --- (13 headers 11 lines)--- Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received INVITE (5) - Command in SIP INVITE Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1004 parse_sip_options: Begin: parsing SIP "Supported: timer,replaces" Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1016 parse_sip_options: Found SIP option: -timer- Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1022 parse_sip_options: Matched SIP option: timer Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1016 parse_sip_options: Found SIP option: -replaces- Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1022 parse_sip_options: Matched SIP option: replaces Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1033 parse_sip_options: * SIP extension value: 5 for call 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 Using INVITE request as basis request - 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 Sending to 192.168.0.50 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:8000 Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 0.0.0.0:8000 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0x1e (gsm|ulaw|alaw|g726), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 13 19:30:33 DEBUG[17674]: channel.c:2350 set_format: Set channel SIP/200-6a41 to write format slin -- Started music on hold, class 'default', on channel 'SIP/200-6a41' Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:10551 handle_request_invite: Got a SIP re-invite for call 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 We're at 192.168.0.160 port 18438 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T3CC17349;received=192.168.0.50 From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200156 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 17655 17656 IN IP4 192.168.0.160 s=session c=IN IP4 192.168.0.160 t=0 0 m=audio 18438 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 13 19:30:33 DEBUG[17674]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #42 Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #42 (1) SIP/2.0 - 1 Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 56 ms (t1 28 ms (Retrans id #42)) Retransmitting #1 (no NAT) to 192.168.0.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T3CC17349;received=192.168.0.50 From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200156 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 17655 17656 IN IP4 192.168.0.160 s=session c=IN IP4 192.168.0.160 t=0 0 m=audio 18438 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.0.50:5060: ACK sip:200@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T7C3C7CFF From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200156 ACK User-Agent: ALL7950 02.09.26 Contact: Max-Forwards: 70 Content-Length: 0 Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: ACK sip:200@192.168.0.160 SIP/2.0 (33) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T7C3C7CFF (72) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=000FC9015027_T2012322891 (62) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: "200" ;tag=as4dd76d35 (48) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 1391200156 ACK (20) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: ALL7950 02.09.26 (28) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Contact: (36) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #42 Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160' of Response 1391200156: Match Found Jul 13 19:30:34 DEBUG[17696]: rtp.c:1260 ast_rtp_raw_write: Difference is 776, ms is 117 <-- SIP read from 192.168.0.50:5060: ACK sip:200@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T083D44A3 From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200156 ACK User-Agent: ALL7950 02.09.26 Contact: Max-Forwards: 70 Content-Length: 0 Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: ACK sip:200@192.168.0.160 SIP/2.0 (33) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T083D44A3 (72) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=000FC9015027_T2012322891 (62) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: "200" ;tag=as4dd76d35 (48) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 1391200156 ACK (20) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: ALL7950 02.09.26 (28) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Contact: (36) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (1) REGISTER - 2 Jul 13 19:30:34 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #41)) Retransmitting #1 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK79cb4c6b;rport From: ;tag=as43326fea To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:35 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (2) REGISTER - 2 Jul 13 19:30:35 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #41)) Retransmitting #2 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK79cb4c6b;rport From: ;tag=as43326fea To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:37 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (3) REGISTER - 2 Jul 13 19:30:37 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #41)) Retransmitting #3 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK79cb4c6b;rport From: ;tag=as43326fea To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 217.10.79.9:5060: --- (0 headers 0 lines) Nat keepalive --- <-- SIP read from 82.165.39.16:5060: --- (0 headers 0 lines) Nat keepalive --- Jul 13 19:30:41 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (4) REGISTER - 2 Jul 13 19:30:41 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #41)) Retransmitting #4 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK79cb4c6b;rport From: ;tag=as43326fea To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- Jul 13 19:30:41 DEBUG[17674]: chan_sip.c:1323 __sip_autodestruct: Auto destroying call '718f233b7a1f8f08321df0d82be766d0@127.0.0.2' Destroying call '718f233b7a1f8f08321df0d82be766d0@127.0.0.2' <-- SIP read from 192.168.0.50:5060: INVITE sip:200@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T36B2C553 From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200157 INVITE Contact: Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Supported: timer,replaces User-Agent: ALL7950 02.09.26 Content-Type: application/sdp Content-Length: 218 v=0 o=100 290009649 290009651 IN IP4 192.168.0.50 s=ALL7950 02.09.26 c=IN IP4 192.168.0.50 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: INVITE sip:200@192.168.0.160 SIP/2.0 (36) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T36B2C553 (72) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=000FC9015027_T2012322891 (62) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: "200" ;tag=as4dd76d35 (48) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 1391200157 INVITE (23) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: Contact: (36) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Max-Forwards: 70 (16) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO (74) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Supported: timer,replaces (25) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: User-Agent: ALL7950 02.09.26 (28) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 11: Content-Type: application/sdp (29) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 12: Content-Length: 218 (19) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 13: (0) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: v=0 (3) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: o=100 290009649 290009651 IN IP4 192.168.0.50 (45) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: s=ALL7950 02.09.26 (18) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: c=IN IP4 192.168.0.50 (21) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: t=0 0 (5) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: m=audio 8000 RTP/AVP 8 101 (26) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:8 PCMA/8000/1 (22) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=fmtp:101 0-15 (15) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3395 parse_request: Line: a=sendrecv (10) --- (13 headers 10 lines)--- Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 Sending to 192.168.0.50 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.50:8000 Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3604 process_sdp: Peer audio RTP is at port 192.168.0.50:8000 Found description format PCMA Found description format telephone-event Capabilities: us - 0x1e (gsm|ulaw|alaw|g726), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 13 19:30:43 DEBUG[17674]: channel.c:2350 set_format: Set channel SIP/200-6a41 to write format alaw -- Stopped music on hold on SIP/200-6a41 Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:10551 handle_request_invite: Got a SIP re-invite for call 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 We're at 192.168.0.160 port 18438 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T36B2C553;received=192.168.0.50 From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200157 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 17655 17657 IN IP4 192.168.0.160 s=session c=IN IP4 192.168.0.160 t=0 0 m=audio 18438 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #43 <-- SIP read from 192.168.0.50:5060: ACK sip:200@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T5789AA99 From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200157 ACK User-Agent: ALL7950 02.09.26 Contact: Max-Forwards: 70 Content-Length: 0 Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: ACK sip:200@192.168.0.160 SIP/2.0 (33) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T5789AA99 (72) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=000FC9015027_T2012322891 (62) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: "200" ;tag=as4dd76d35 (48) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 1391200157 ACK (20) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: ALL7950 02.09.26 (28) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Contact: (36) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #43 Jul 13 19:30:43 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160' of Response 1391200157: Match Found Jul 13 19:30:43 DEBUG[17667]: chan_iax2.c:7657 iax2_do_register: Allocate call number Jul 13 19:30:43 DEBUG[17667]: chan_iax2.c:7663 iax2_do_register: Registration created on call 9 Jul 13 19:30:45 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (5) REGISTER - 2 Jul 13 19:30:45 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #41)) Retransmitting #5 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK79cb4c6b;rport From: ;tag=as43326fea To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 192.168.0.50:5060: BYE sip:200@192.168.0.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T704E5CE5 From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200158 BYE User-Agent: ALL7950 02.09.26 Contact: Max-Forwards: 70 Content-Length: 0 Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: BYE sip:200@192.168.0.160 SIP/2.0 (33) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T704E5CE5 (72) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=000FC9015027_T2012322891 (62) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: "200" ;tag=as4dd76d35 (48) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 (55) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 1391200158 BYE (20) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: ALL7950 02.09.26 (28) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Contact: (36) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: Max-Forwards: 70 (16) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 9: Content-Length: 0 (17) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:11137 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 192.168.0.50 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK_000FC9015027_T704E5CE5;received=192.168.0.50 From: ;tag=000FC9015027_T2012322891 To: "200" ;tag=as4dd76d35 Call-ID: 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160 CSeq: 1391200158 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Jul 13 19:30:47 DEBUG[17696]: channel.c:3275 ast_generic_bridge: Didn't get a frame from channel: SIP/100-3e19 Jul 13 19:30:47 DEBUG[17696]: channel.c:3550 ast_channel_bridge: Bridge stops bridging channels SIP/200-6a41 and SIP/100-3e19 Jul 13 19:30:47 DEBUG[17696]: channel.c:1323 ast_hangup: Hanging up channel 'SIP/100-3e19' Jul 13 19:30:47 DEBUG[17696]: chan_sip.c:2418 sip_hangup: Hangup call SIP/100-3e19, SIP callid 685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160) Jul 13 19:30:47 DEBUG[17696]: chan_sip.c:2426 sip_hangup: update_call_counter(100) - decrement call limit counter Jul 13 19:30:47 DEBUG[17696]: chan_sip.c:2209 update_call_counter: Updating call counter for incoming call Jul 13 19:30:47 DEBUG[17696]: app_dial.c:1619 dial_exec_full: Exiting with DIALSTATUS=ANSWER. Jul 13 19:30:47 DEBUG[17696]: pbx.c:2316 __ast_pbx_run: Spawn extension (Softphone,100,2) exited non-zero on 'SIP/200-6a41' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1677 pbx_extension_helper: Launching 'NoOp' -- Executing NoOp("SIP/200-6a41", "Zeitstempel: 13072006-19:30:47 ") in new stack Jul 13 19:30:47 DEBUG[17696]: pbx.c:1677 pbx_extension_helper: Launching 'NoOp' -- Executing NoOp("SIP/200-6a41", "Hangup Event Nebenstelle.: "200" <200>") in new stack Jul 13 19:30:47 DEBUG[17696]: pbx.c:1677 pbx_extension_helper: Launching 'Hangup' -- Executing Hangup("SIP/200-6a41", "") in new stack Jul 13 19:30:47 DEBUG[17696]: pbx.c:2454 __ast_pbx_run: Spawn extension (Softphone,h,3) exited non-zero on 'SIP/200-6a41' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '"200" <200>' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '200' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '100' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Softphone' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/200-6a41' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/100-3e19' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Hangup' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-13 19:30:10' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-13 19:30:13' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-13 19:30:47' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '37' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '34' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'IP-Telefon-2' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1152811810.0' Jul 13 19:30:47 DEBUG[17696]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 13 19:30:47 DEBUG[17696]: channel.c:1323 ast_hangup: Hanging up channel 'SIP/200-6a41' Jul 13 19:30:47 DEBUG[17696]: chan_sip.c:2418 sip_hangup: Hangup call SIP/200-6a41, SIP callid CALL_ID9_000FC9014CE3_T325875288@192.168.0.41) Jul 13 19:30:47 DEBUG[17696]: chan_sip.c:2426 sip_hangup: update_call_counter(200) - decrement call limit counter Jul 13 19:30:47 DEBUG[17696]: chan_sip.c:2209 update_call_counter: Updating call counter for incoming call Destroying call '685aa2f345dd62754c50bf3f0530b3d8@192.168.0.160' Jul 13 19:30:47 DEBUG[17659]: chan_sip.c:11668 sip_devicestate: Checking device state for peer 100 Jul 13 19:30:47 DEBUG[17659]: devicestate.c:187 do_state_change: Changing state for SIP/100 - state 1 (Not in use) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.41, port 5060 Reliably Transmitting (no NAT) to 192.168.0.41:5060: BYE sip:200@192.168.0.41:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK32e51a1a;rport From: ;tag=as018c72c1 To: "200" ;tag=000FC9014CE3_T274886762 Contact: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 13 19:30:47 DEBUG[17696]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #44 Jul 13 19:30:47 DEBUG[17701]: app_queue.c:523 changethread: Device 'SIP/100' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Jul 13 19:30:47 DEBUG[17659]: chan_sip.c:11668 sip_devicestate: Checking device state for peer 200 Jul 13 19:30:47 DEBUG[17659]: devicestate.c:187 do_state_change: Changing state for SIP/200 - state 1 (Not in use) Jul 13 19:30:47 DEBUG[17702]: app_queue.c:523 changethread: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <-- SIP read from 192.168.0.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK32e51a1a;rport From: ;tag=as018c72c1 To: "200" ;tag=000FC9014CE3_T274886762 Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 CSeq: 102 BYE User-Agent: ALL7950 02.09.18 Content-Length: 0 Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 0: SIP/2.0 200 OK (14) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK32e51a1a;rport (64) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 2: From: ;tag=as018c72c1 (49) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 3: To: "200" ;tag=000FC9014CE3_T274886762 (66) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 4: Call-ID: CALL_ID9_000FC9014CE3_T325875288@192.168.0.41 (54) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 5: CSeq: 102 BYE (13) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 6: User-Agent: ALL7950 02.09.18 (28) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 7: Content-Length: 0 (17) Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:3363 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #44 Jul 13 19:30:47 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on 'CALL_ID9_000FC9014CE3_T325875288@192.168.0.41' of Request 102: Match Found Destroying call 'CALL_ID9_000FC9014CE3_T325875288@192.168.0.41' Jul 13 19:30:49 DEBUG[17674]: chan_sip.c:1184 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (6) REGISTER - 2 Jul 13 19:30:49 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #41)) Retransmitting #6 (no NAT) to 82.139.223.1:5060: REGISTER sip:sip.sipport.de SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK79cb4c6b;rport From: ;tag=as43326fea To: Call-ID: 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 1200 Contact: Event: registration Content-Length: 0 --- sip no debug SIP Debugging Disabled *CLI> Jul 13 19:30:53 DEBUG[17667]: chan_iax2.c:7163 socket_read: Peer lastms 2, historicms 2, maxms 2000 Jul 13 19:30:53 DEBUG[17667]: chan_iax2.c:7163 socket_read: Peer lastms 13, historicms 13, maxms 2000 Jul 13 19:30:53 DEBUG[17667]: chan_iax2.c:7163 socket_read: Peer lastms 22, historicms 18, maxms 2000 Jul 13 19:30:53 DEBUG[17667]: chan_iax2.c:7163 socket_read: Peer lastms 29, historicms 24, maxms 2000 Jul 13 19:30:53 DEBUG[17674]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Jul 13 19:30:53 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '2bdbe528061cdcbc33a8f25e63d92302@192.168.0.160' of Request 102: Match Found Jul 13 19:30:53 DEBUG[17674]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Jul 13 19:30:53 DEBUG[17674]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Jul 13 19:30:53 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '43265e7449ed636027fd299410bdb627@192.168.0.160' of Request 102: Match Found Jul 13 19:30:53 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '4d5ff08872f9e5b726644d1864a08feb@192.168.0.160' of Request 102: Match Found Jul 13 19:30:53 NOTICE[17674]: chan_sip.c:5372 sip_reg_timeout: -- Registration for 'sip22660@sip.sipport.de' timed out, trying again (Attempt #3) Jul 13 19:30:53 DEBUG[17674]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '2859f57d4fd8882647603bc55c4536b5@127.0.0.2' of Request 104: Match Found Jul 13 19:30:53 DEBUG[17674]: chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for 2859f57d4fd8882647603bc55c4536b5@127.0.0.2 - REGISTER (No RTP) Jul 13 19:30:53 DEBUG[17674]: chan_sip.c:5507 transmit_register: Scheduled a registration timeout for sip.sipport.de id #54 REGISTER attempt 4 to sip22660@sip.sipport.de Jul 13 19:30:54 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #55)) set vJul 13 19:30:55 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #55)) erbose 0 Verbosity is now OFF *CLI> set debJul 13 19:30:57 DEBUG[17674]: chan_sip.c:1198 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #55)) ug 0 Core debug is now OFF