*CLI> sip debug peer 1000 SIP Debugging Enabled for IP: 192.168.62.101:5060 *CLI> Mar 6 23:11:49 DEBUG[26172]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '174063205b190351411eb7f1681f832d@192.168.64.1' Mar 6 23:11:49 DEBUG[26172]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '09d57bfc64ed8ae7200941c72cc66307@213.84.46.114' Mar 6 23:11:49 DEBUG[26172]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '0f776760715a6f19422b39700f075ca2@192.168.64.1' Destroying call '0f776760715a6f19422b39700f075ca2@192.168.64.1' -- Starting simple switch on 'Zap/1-1' Mar 6 23:12:06 DEBUG[26172]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '06b2df595a0952bd7378be2324386acd@192.168.64.1' Mar 6 23:12:09 NOTICE[26306]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Answer("Zap/1-1", "") in new stack Mar 6 23:12:09 DEBUG[26306]: chan_zap.c:2677 zt_answer: Took Zap/1-1 off hook Mar 6 23:12:09 DEBUG[26306]: chan_zap.c:1405 zt_enable_ec: Enabled echo cancellation on channel 1 Mar 6 23:12:09 DEBUG[26306]: chan_zap.c:1424 zt_train_ec: No echo training requested -- Executing Wait("Zap/1-1", "1") in new stack Mar 6 23:12:09 DEBUG[26164]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'Zap/1-1' -- Executing SetMusicOnHold("Zap/1-1", "default") in new stack -- Executing BackGround("Zap/1-1", "transfer") in new stack Mar 6 23:12:10 DEBUG[26306]: channel.c:1713 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'transfer' (language 'en') Mar 6 23:12:16 DEBUG[26306]: channel.c:1713 ast_settimeout: Scheduling timer at 56 sample intervals Mar 6 23:12:16 DEBUG[26306]: channel.c:1713 ast_settimeout: Scheduling timer at 0 sample intervals Mar 6 23:12:16 DEBUG[26306]: channel.c:1713 ast_settimeout: Scheduling timer at 0 sample intervals -- Executing ResponseTimeout("Zap/1-1", "10") in new stack Mar 6 23:12:16 WARNING[26306]: pbx.c:5804 pbx_builtin_rtimeout: ResponseTimeout is deprecated, please use Set(TIMEOUT(response)=timeout) instead. -- Set Response Timeout to 10 -- Executing Queue("Zap/1-1", "callqueue|ntTH|||20") in new stack -- Started music on hold, class 'default', on channel 'Zap/1-1' Mar 6 23:12:16 DEBUG[26306]: channel.c:1713 ast_settimeout: Scheduling timer at 160 sample intervals Mar 6 23:12:16 DEBUG[26306]: chan_sip.c:1864 create_addr_from_peer: Setting NAT on RTP to 524288 Mar 6 23:12:16 DEBUG[26306]: chan_sip.c:2058 sip_call: Outgoing Call for -- Called SIP/1002 Mar 6 23:12:16 DEBUG[26306]: chan_sip.c:1864 create_addr_from_peer: Setting NAT on RTP to 524288 Mar 6 23:12:16 DEBUG[26306]: chan_sip.c:2058 sip_call: Outgoing Call for -- Called SIP/1001 Mar 6 23:12:16 DEBUG[26306]: chan_sip.c:1864 create_addr_from_peer: Setting NAT on RTP to 524288 Mar 6 23:12:16 DEBUG[26306]: chan_sip.c:2058 sip_call: Outgoing Call for We're at 192.168.64.1 port 19984 Adding codec 0x40 (slin) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726) to SDP Adding codec 0x20 (adpcm) 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=z9hG4bK73c7f8ea;rport From: "asterisk" ;tag=as66f51fe5 To: Contact: Call-ID: 4200166972d09aae4799c855516f4451@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 06 Mar 2006 22:12:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 492 v=0 o=root 26306 26306 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19984 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101 a=rtpmap:10 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 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 6 23:12:16 DEBUG[26306]: channel.c:1975 ast_read: Generator got voice, switching to phase locked mode Mar 6 23:12:16 DEBUG[26306]: 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=z9hG4bK73c7f8ea;rport From: "asterisk" ;tag=as66f51fe5 To: Contact: Call-ID: 4200166972d09aae4799c855516f4451@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 06 Mar 2006 22:12:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 492 v=0 o=root 26306 26306 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19984 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101 a=rtpmap:10 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 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=z9hG4bK73c7f8ea;rport From: "asterisk" ;tag=as66f51fe5 To: Contact: Call-ID: 4200166972d09aae4799c855516f4451@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 06 Mar 2006 22:12:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 492 v=0 o=root 26306 26306 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19984 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101 a=rtpmap:10 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 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=z9hG4bK73c7f8ea;rport From: "asterisk" ;tag=as66f51fe5 To: Contact: Call-ID: 4200166972d09aae4799c855516f4451@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 06 Mar 2006 22:12:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 492 v=0 o=root 26306 26306 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19984 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101 a=rtpmap:10 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 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=z9hG4bK73c7f8ea;rport From: "asterisk" ;tag=as66f51fe5 To: Contact: Call-ID: 4200166972d09aae4799c855516f4451@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 06 Mar 2006 22:12:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 492 v=0 o=root 26306 26306 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19984 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101 a=rtpmap:10 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 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 #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=z9hG4bK73c7f8ea;rport From: "asterisk" ;tag=as66f51fe5 To: Contact: Call-ID: 4200166972d09aae4799c855516f4451@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 06 Mar 2006 22:12:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 492 v=0 o=root 26306 26306 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19984 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101 a=rtpmap:10 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 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=z9hG4bK73c7f8ea;rport From: "asterisk" ;tag=as66f51fe5 To: Contact: Call-ID: 4200166972d09aae4799c855516f4451@192.168.64.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 06 Mar 2006 22:12:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 492 v=0 o=root 26306 26306 IN IP4 192.168.64.1 s=session c=IN IP4 192.168.64.1 t=0 0 m=audio 19984 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101 a=rtpmap:10 L16/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 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 6 23:12:52 DEBUG[26306]: chan_zap.c:4351 __zt_exception: Exception on 18, channel 1 Mar 6 23:12:52 DEBUG[26306]: chan_zap.c:3539 zt_handle_event: Got event On hook(1) on channel 1 (index 0) Mar 6 23:12:52 DEBUG[26306]: chan_zap.c:1437 zt_disable_ec: disabled echo cancellation on channel 1 -- Stopped music on hold on Zap/1-1 Mar 6 23:12:52 DEBUG[26306]: channel.c:1713 ast_settimeout: Scheduling timer at 0 sample intervals Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:2416 sip_hangup: update_call_counter() - decrement call limit counter Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:1372 __sip_ack: Acked pending invite 102 Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '32ac17db54d050521a6377f364f53093@192.168.64.1' of Request 102: Match Found Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '32ac17db54d050521a6377f364f53093@192.168.64.1' of Request 102: Match Found Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:2416 sip_hangup: update_call_counter() - decrement call limit counter Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:1372 __sip_ack: Acked pending invite 102 Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '124ccb9c3d013c331a3a980058538a98@213.84.46.114' of Request 102: Match Found Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '124ccb9c3d013c331a3a980058538a98@213.84.46.114' of Request 102: Match Found Mar 6 23:12:52 DEBUG[26306]: 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=z9hG4bK73c7f8ea;rport From: "asterisk" ;tag=as66f51fe5 To: Contact: Call-ID: 4200166972d09aae4799c855516f4451@192.168.64.1 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '4200166972d09aae4799c855516f4451@192.168.64.1' in 15000 ms Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:1372 __sip_ack: Acked pending invite 102 Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '4200166972d09aae4799c855516f4451@192.168.64.1' of Request 102: Match Found Mar 6 23:12:52 DEBUG[26306]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '4200166972d09aae4799c855516f4451@192.168.64.1' of Request 102: Match Found == Spawn extension (bell, s, 6) exited non-zero on 'Zap/1-1' Mar 6 23:12:52 DEBUG[26306]: chan_zap.c:2345 zt_hangup: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Mar 6 23:12:52 DEBUG[26306]: chan_zap.c:1437 zt_disable_ec: disabled echo cancellation on channel 1 Mar 6 23:12:52 DEBUG[26306]: chan_zap.c:2785 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1 Mar 6 23:12:52 DEBUG[26306]: chan_zap.c:1374 update_conf: Updated conferencing on 1, with 0 conference users -- Hungup 'Zap/1-1' Mar 6 23:13:07 DEBUG[26172]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '32ac17db54d050521a6377f364f53093@192.168.64.1' Mar 6 23:13:07 DEBUG[26172]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '124ccb9c3d013c331a3a980058538a98@213.84.46.114' Mar 6 23:13:07 DEBUG[26172]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '4200166972d09aae4799c855516f4451@192.168.64.1' Destroying call '4200166972d09aae4799c855516f4451@192.168.64.1' Mar 6 23:13:19 NOTICE[26172]: chan_sip.c:5239 sip_reregister: -- Re-registration for 0878710766@sip.xs4all.nl Mar 6 23:13:19 DEBUG[26172]: chan_sip.c:5402 transmit_register: Scheduled a registration timeout for sip.xs4all.nl id #22 Mar 6 23:13:19 DEBUG[26172]: chan_sip.c:5453 transmit_register: >>> Re-using Auth data for 0878710766@sip.xs4all.nl Mar 6 23:13:19 DEBUG[26172]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '06b2df595a0952bd7378be2324386acd@192.168.64.1' of Request 104: Match Found Mar 6 23:13:19 DEBUG[26172]: chan_sip.c:1447 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '06b2df595a0952bd7378be2324386acd@192.168.64.1' Request 105: Found Mar 6 23:13:19 DEBUG[26172]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '06b2df595a0952bd7378be2324386acd@192.168.64.1' of Request 105: Match Found Mar 6 23:13:19 DEBUG[26172]: chan_sip.c:9614 handle_response_register: Registration successful Mar 6 23:13:19 DEBUG[26172]: chan_sip.c:9616 handle_response_register: Cancelling timeout 22 Mar 6 23:13:19 NOTICE[26172]: chan_sip.c:9666 handle_response_register: Outbound Registration: Expiry for sip.xs4all.nl is 120 sec (Scheduling reregistration in 105 s)