*CLI> sip debug peer 1000 SIP Debugging Enabled for IP: 192.168.62.101:5060 *CLI> *CLI> *CLI> Mar 7 20:23:36 DEBUG[3682]: chan_sip.c:7101 check_user_full: Setting NAT on RTP to 524288 Mar 7 20:23:36 DEBUG[3682]: chan_sip.c:10327 handle_request_invite: Checking SIP call limits for device 0878710766 Mar 7 20:23:36 DEBUG[3682]: chan_sip.c:6025 build_route: build_route: Contact hop: -- Executing Answer("SIP/0878710766-84b4", "") in new stack -- Executing Wait("SIP/0878710766-84b4", "1") in new stack Mar 7 20:23:36 DEBUG[3682]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '03-01391-04ae2760-0890cf315@sip.xs4all.nl' of Response 63053385: Match Found -- Executing SetMusicOnHold("SIP/0878710766-84b4", "default") in new stack -- Executing BackGround("SIP/0878710766-84b4", "transfer") in new stack Mar 7 20:23:37 DEBUG[4043]: channel.c:1713 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'transfer' (language 'en') Mar 7 20:23:38 WARNING[3682]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 45e972f177018db549fa14f96d32ff19@192.168.64.1 for seqno 102 (Critical Request) Mar 7 20:23:38 WARNING[3682]: chan_sip.c:1227 retrans_pkt: Hanging up call 45e972f177018db549fa14f96d32ff19@192.168.64.1 - no reply to our critical packet.Mar 7 20:23:38 WARNING[3682]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 3a9c05042afa75575bb6e4e95ce3654f@213.84.46.114 for seqno 102 (Critical Request) Mar 7 20:23:38 WARNING[3682]: chan_sip.c:1227 retrans_pkt: Hanging up call 3a9c05042afa75575bb6e4e95ce3654f@213.84.46.114 - no reply to our critical packet. Mar 7 20:23:38 WARNING[3682]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 1bdfefd2362f0da36991d9d442848610@192.168.64.1 for seqno 102 (Critical Request) Mar 7 20:23:38 WARNING[3682]: chan_sip.c:1227 retrans_pkt: Hanging up call 1bdfefd2362f0da36991d9d442848610@192.168.64.1 - no reply to our critical packet.Mar 7 20:23:38 DEBUG[3734]: chan_sip.c:2416 sip_hangup: update_call_counter() - decrement call limit counter Mar 7 20:23:38 DEBUG[3734]: chan_sip.c:2416 sip_hangup: update_call_counter() - decrement call limit counter Mar 7 20:23:38 DEBUG[3734]: chan_sip.c:2416 sip_hangup: update_call_counter() - decrement call limit counter Mar 7 20:23:38 NOTICE[3734]: app_queue.c:1662 wait_for_answer: No one is answering queue 'callqueue' (3/0/0) -- Stopped music on hold on Zap/1-1 Mar 7 20:23:38 DEBUG[3734]: channel.c:1713 ast_settimeout: Scheduling timer at 0 sample intervals -- Executing BackGround("Zap/1-1", "leavemsg") in new stack Mar 7 20:23:38 WARNING[3734]: file.c:509 ast_openstream_full: File leavemsg does not exist in any format Mar 7 20:23:38 WARNING[3734]: file.c:821 ast_streamfile: Unable to open leavemsg (format unknown): No such file or directory Mar 7 20:23:38 WARNING[3734]: pbx.c:5757 pbx_builtin_background: ast_streamfile failed on Zap/1-1 for leavemsg -- Executing VoiceMail("Zap/1-1", "u0000") in new stack Mar 7 20:23:38 WARNING[3734]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '0000' -- Executing Hangup("Zap/1-1", "") in new stack == Spawn extension (bell, s, 9) exited non-zero on 'Zap/1-1' Mar 7 20:23:38 DEBUG[3734]: chan_zap.c:2345 zt_hangup: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Mar 7 20:23:38 DEBUG[3734]: chan_zap.c:1437 zt_disable_ec: disabled echo cancellation on channel 1 Mar 7 20:23:38 DEBUG[3734]: chan_zap.c:2785 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1 Mar 7 20:23:38 DEBUG[3734]: chan_zap.c:1374 update_conf: Updated conferencing on 1, with 0 conference users -- Hungup 'Zap/1-1' Destroying call '1bdfefd2362f0da36991d9d442848610@192.168.64.1' Mar 7 20:23:43 DEBUG[4043]: channel.c:1713 ast_settimeout: Scheduling timer at 56 sample intervals Mar 7 20:23:43 DEBUG[4043]: channel.c:1713 ast_settimeout: Scheduling timer at 0 sample intervals Mar 7 20:23:43 DEBUG[4043]: channel.c:1713 ast_settimeout: Scheduling timer at 0 sample intervals -- Executing ResponseTimeout("SIP/0878710766-84b4", "10") in new stack -- Set Response Timeout to 10 -- Executing Queue("SIP/0878710766-84b4", "callqueue|ntTH|||20") in new stack -- Started music on hold, class 'default', on channel 'SIP/0878710766-84b4' Mar 7 20:23:43 DEBUG[4043]: channel.c:1713 ast_settimeout: Scheduling timer at 160 sample intervals Mar 7 20:23:43 DEBUG[4043]: chan_sip.c:1864 create_addr_from_peer: Setting NAT on RTP to 524288 Mar 7 20:23:43 DEBUG[4043]: chan_sip.c:2058 sip_call: Outgoing Call for -- Called SIP/1002 Mar 7 20:23:43 DEBUG[4043]: chan_sip.c:1864 create_addr_from_peer: Setting NAT on RTP to 524288 Mar 7 20:23:43 DEBUG[4043]: chan_sip.c:2058 sip_call: Outgoing Call for -- Called SIP/1001 Mar 7 20:23:43 DEBUG[4043]: chan_sip.c:1864 create_addr_from_peer: Setting NAT on RTP to 524288 Mar 7 20:23:43 DEBUG[4043]: chan_sip.c:2058 sip_call: Outgoing Call for We're at 192.168.64.1 port 19966 Adding codec 0x4 (ulaw) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 21 lines Reliably Transmitting (NAT) to 192.168.62.101:5060: INVITE sip:192.168.62.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.64.1:5060;branch=z9hG4bK717d4330;rport From: "0641278122" ;tag=as74159423 To: Contact: Call-ID: 68dd172a773727454db901a8725b2f13@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 07 Mar 2006 19:23:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 4043 4043 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19966 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called SIP/1000 Mar 7 20:23:43 DEBUG[4043]: channel.c:1975 ast_read: Generator got voice, switching to phase locked mode Mar 7 20:23:43 DEBUG[4043]: channel.c:1713 ast_settimeout: Scheduling timer at 0 sample intervals Retransmitting #1 (NAT) to 192.168.62.101:5060: INVITE sip:192.168.62.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.64.1:5060;branch=z9hG4bK717d4330;rport From: "0641278122" ;tag=as74159423 To: Contact: Call-ID: 68dd172a773727454db901a8725b2f13@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 07 Mar 2006 19:23:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 4043 4043 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19966 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (NAT) to 192.168.62.101:5060: INVITE sip:192.168.62.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.64.1:5060;branch=z9hG4bK717d4330;rport From: "0641278122" ;tag=as74159423 To: Contact: Call-ID: 68dd172a773727454db901a8725b2f13@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 07 Mar 2006 19:23:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 4043 4043 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19966 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (NAT) to 192.168.62.101:5060: INVITE sip:192.168.62.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.64.1:5060;branch=z9hG4bK717d4330;rport From: "0641278122" ;tag=as74159423 To: Contact: Call-ID: 68dd172a773727454db901a8725b2f13@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 07 Mar 2006 19:23:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 4043 4043 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19966 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #4 (NAT) to 192.168.62.101:5060: INVITE sip:192.168.62.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.64.1:5060;branch=z9hG4bK717d4330;rport From: "0641278122" ;tag=as74159423 To: Contact: Call-ID: 68dd172a773727454db901a8725b2f13@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 07 Mar 2006 19:23:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 4043 4043 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19966 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Mar 7 20:23:56 NOTICE[3682]: chan_sip.c:5239 sip_reregister: -- Re-registration for 0878710766@sip.xs4all.nl Mar 7 20:23:56 DEBUG[3682]: chan_sip.c:5402 transmit_register: Scheduled a registration timeout for sip.xs4all.nl id #20 Mar 7 20:23:56 DEBUG[3682]: chan_sip.c:5453 transmit_register: >>> Re-using Auth data for 0878710766@sip.xs4all.nl Mar 7 20:23:56 DEBUG[3682]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '10920284495d4ef338a5e43827003a0d@192.168.64.1' of Request 104: Match Found Mar 7 20:23:56 DEBUG[3682]: chan_sip.c:1447 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '10920284495d4ef338a5e43827003a0d@192.168.64.1' Request 105: Found Mar 7 20:23:56 DEBUG[3682]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '10920284495d4ef338a5e43827003a0d@192.168.64.1' of Request 105: Match Found Mar 7 20:23:56 DEBUG[3682]: chan_sip.c:9614 handle_response_register: Registration successful Mar 7 20:23:56 DEBUG[3682]: chan_sip.c:9616 handle_response_register: Cancelling timeout 20 Mar 7 20:23:56 NOTICE[3682]: chan_sip.c:9666 handle_response_register: Outbound Registration: Expiry for sip.xs4all.nl is 120 sec (Scheduling reregistration in 105 s) Mar 7 20:23:57 DEBUG[3684]: chan_iax2.c:4902 raw_hangup: Raw Hangup 127.0.0.1:47346, src=0, dst=0 Retransmitting #5 (NAT) to 192.168.62.101:5060: INVITE sip:192.168.62.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.64.1:5060;branch=z9hG4bK717d4330;rport From: "0641278122" ;tag=as74159423 To: Contact: Call-ID: 68dd172a773727454db901a8725b2f13@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 07 Mar 2006 19:23:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 4043 4043 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19966 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #6 (NAT) to 192.168.62.101:5060: INVITE sip:192.168.62.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.64.1:5060;branch=z9hG4bK717d4330;rport From: "0641278122" ;tag=as74159423 To: Contact: Call-ID: 68dd172a773727454db901a8725b2f13@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 07 Mar 2006 19:23:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 4043 4043 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19966 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Stopped music on hold on SIP/0878710766-84b4 Mar 7 20:24:16 DEBUG[4043]: channel.c:1713 ast_settimeout: Scheduling timer at 0 sample intervals Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:2416 sip_hangup: update_call_counter() - decrement call limit counter Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:1372 __sip_ack: Acked pending invite 102 Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '161a7f245d5dc6765d46f20104305798@192.168.64.1' of Request 102: Match Found Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '161a7f245d5dc6765d46f20104305798@192.168.64.1' of Request 102: Match Found Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:2416 sip_hangup: update_call_counter() - decrement call limit counter Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:1372 __sip_ack: Acked pending invite 102 Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '2c8afbb81c5277bb194bbaa123c01ab8@213.84.46.114' of Request 102: Match Found Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '2c8afbb81c5277bb194bbaa123c01ab8@213.84.46.114' of Request 102: Match Found Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:2416 sip_hangup: update_call_counter() - decrement call limit counter Reliably Transmitting (NAT) to 192.168.62.101:5060: CANCEL sip:192.168.62.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.64.1:5060;branch=z9hG4bK717d4330;rport From: "0641278122" ;tag=as74159423 To: Contact: Call-ID: 68dd172a773727454db901a8725b2f13@192.168.64.1 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '68dd172a773727454db901a8725b2f13@192.168.64.1' in 15000 ms Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:1372 __sip_ack: Acked pending invite 102 Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '68dd172a773727454db901a8725b2f13@192.168.64.1' of Request 102: Match Found Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '68dd172a773727454db901a8725b2f13@192.168.64.1' of Request 102: Match Found == Spawn extension (tcg, tcg, 6) exited non-zero on 'SIP/0878710766-84b4' Mar 7 20:24:16 DEBUG[4043]: chan_sip.c:2416 sip_hangup: update_call_counter(0878710766) - decrement call limit counter Mar 7 20:24:28 DEBUG[3682]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '10920284495d4ef338a5e43827003a0d@192.168.64.1' Mar 7 20:24:31 DEBUG[3682]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '161a7f245d5dc6765d46f20104305798@192.168.64.1' Mar 7 20:24:31 DEBUG[3682]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '2c8afbb81c5277bb194bbaa123c01ab8@213.84.46.114' Mar 7 20:24:31 DEBUG[3682]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '68dd172a773727454db901a8725b2f13@192.168.64.1' Destroying call '68dd172a773727454db901a8725b2f13@192.168.64.1'