[Mar 21 11:07:03] VERBOSE[6211] logger.c: Asterisk Event Logger restarted [Mar 21 11:07:03] VERBOSE[6211] logger.c: Asterisk Queue Logger restarted [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Auto destroying SIP dialog '5a91a193-18404649@192.168.1.186' [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Destroying SIP dialog 5a91a193-18404649@192.168.1.186 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Really destroying SIP dialog '5a91a193-18404649@192.168.1.186' Method: REGISTER [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Auto destroying SIP dialog 'fdfd1531-acfb3743@192.168.1.186' [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Destroying SIP dialog fdfd1531-acfb3743@192.168.1.186 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Really destroying SIP dialog 'fdfd1531-acfb3743@192.168.1.186' Method: REGISTER [Mar 21 11:07:07] VERBOSE[6231] logger.c: <--- SIP read from 192.168.1.235:5061 ---> INVITE sip:94006666@192.168.1.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKpgudkhgv Max-Forwards: 70 To: From: "Miguel" ;tag=ssiea Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 275 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE, INFO Supported: 100rel User-Agent: Twinkle/0.9 Content-Length: 306 v=0 o=703 1471312009 412960044 IN IP4 192.168.1.235 s=- c=IN IP4 192.168.1.235 t=0 0 m=audio 8002 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 0: INVITE sip:94006666@192.168.1.235 SIP/2.0 (41) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKpgudkhgv (64) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 3: To: (32) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 4: From: "Miguel" ;tag=ssiea (48) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 5: Call-ID: ttcjtjstxvndzeg@192.168.1.235 (38) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 6: CSeq: 275 INVITE (16) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 7: Contact: (37) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 8: Content-Type: application/sdp (29) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 9: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE, INFO (79) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 10: Supported: 100rel (17) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 11: User-Agent: Twinkle/0.9 (23) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 12: Content-Length: 306 (19) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 13: (0) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: v=0 (3) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: o=703 1471312009 412960044 IN IP4 192.168.1.235 (47) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: s=- (3) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: c=IN IP4 192.168.1.235 (22) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: t=0 0 (5) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: m=audio 8002 RTP/AVP 98 97 8 0 3 101 (36) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:98 speex/16000 (23) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:97 speex/8000 (22) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=ptime:20 (10) [Mar 21 11:07:07] VERBOSE[6231] logger.c: --- (13 headers 14 lines) --- [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Setting NAT on RTP to Off [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Allocating new SIP dialog for ttcjtjstxvndzeg@192.168.1.235 - INVITE (With RTP) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Begin: parsing SIP "Supported: 100rel" [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Found SIP option: -100rel- [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Matched SIP option: 100rel [Mar 21 11:07:07] VERBOSE[6231] logger.c: Sending to 192.168.1.235 : 5061 (NAT) [Mar 21 11:07:07] VERBOSE[6231] logger.c: Using INVITE request as basis request - ttcjtjstxvndzeg@192.168.1.235 [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Setting NAT on RTP to Off [Mar 21 11:07:07] VERBOSE[6231] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.1.235:5061 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.235:5061;branch=z9hG4bKpgudkhgv;received=192.168.1.235;rport=5061 From: "Miguel" ;tag=ssiea To: ;tag=as2fafc884 Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 275 INVITE User-Agent: Netgate Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="00559a27" Content-Length: 0 <------------> [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #179 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Scheduling destruction of SIP dialog 'ttcjtjstxvndzeg@192.168.1.235' in 32000 ms (Method: INVITE) [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found user '703' [Mar 21 11:07:07] VERBOSE[6231] logger.c: <--- SIP read from 192.168.1.235:5061 ---> ACK sip:94006666@192.168.1.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKpgudkhgv Max-Forwards: 70 To: ;tag=as2fafc884 From: "Miguel" ;tag=ssiea Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 275 ACK User-Agent: Twinkle/0.9 Content-Length: 0 <-------------> [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 0: ACK sip:94006666@192.168.1.235 SIP/2.0 (38) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKpgudkhgv (64) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 3: To: ;tag=as2fafc884 (47) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 4: From: "Miguel" ;tag=ssiea (48) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 5: Call-ID: ttcjtjstxvndzeg@192.168.1.235 (38) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 6: CSeq: 275 ACK (13) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 7: User-Agent: Twinkle/0.9 (23) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 8: Content-Length: 0 (17) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 9: (0) [Mar 21 11:07:07] VERBOSE[6231] logger.c: --- (9 headers 0 lines) --- [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #179 [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Stopping retransmission on 'ttcjtjstxvndzeg@192.168.1.235' of Response 275: Match Not Found [Mar 21 11:07:07] VERBOSE[6231] logger.c: <--- SIP read from 192.168.1.235:5061 ---> INVITE sip:94006666@192.168.1.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKepvhqzqj Max-Forwards: 70 Proxy-Authorization: Digest username="703",realm="asterisk",nonce="00559a27",uri="sip:94006666@192.168.1.235",response="d749b296e592f2e87020fb9df7bea852",algorithm=MD5 To: From: "Miguel" ;tag=ssiea Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 276 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE, INFO Supported: 100rel User-Agent: Twinkle/0.9 Content-Length: 306 v=0 o=703 1471312009 412960044 IN IP4 192.168.1.235 s=- c=IN IP4 192.168.1.235 t=0 0 m=audio 8002 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 0: INVITE sip:94006666@192.168.1.235 SIP/2.0 (41) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKepvhqzqj (64) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 3: Proxy-Authorization: Digest username="703",realm="asterisk",nonce="00559a27",uri="sip:94006666@192.168.1.235",response="d749b296e592f2e87020fb9df7bea852",algorithm=MD5 (167) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 4: To: (32) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 5: From: "Miguel" ;tag=ssiea (48) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 6: Call-ID: ttcjtjstxvndzeg@192.168.1.235 (38) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 7: CSeq: 276 INVITE (16) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 8: Contact: (37) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 9: Content-Type: application/sdp (29) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 10: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE, INFO (79) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 11: Supported: 100rel (17) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 12: User-Agent: Twinkle/0.9 (23) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 13: Content-Length: 306 (19) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 14: (0) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: v=0 (3) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: o=703 1471312009 412960044 IN IP4 192.168.1.235 (47) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: s=- (3) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: c=IN IP4 192.168.1.235 (22) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: t=0 0 (5) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: m=audio 8002 RTP/AVP 98 97 8 0 3 101 (36) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:98 speex/16000 (23) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:97 speex/8000 (22) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=ptime:20 (10) [Mar 21 11:07:07] VERBOSE[6231] logger.c: --- (14 headers 14 lines) --- [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Mar 21 11:07:07] VERBOSE[6231] logger.c: Sending to 192.168.1.235 : 5061 (NAT) [Mar 21 11:07:07] VERBOSE[6231] logger.c: Using INVITE request as basis request - ttcjtjstxvndzeg@192.168.1.235 [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Setting NAT on RTP to Off [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found user '703' [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found RTP audio format 98 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found RTP audio format 97 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found RTP audio format 8 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found RTP audio format 0 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found RTP audio format 3 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found RTP audio format 101 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Peer audio RTP is at port 192.168.1.235:8002 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found description format speex for ID 98 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found description format speex for ID 97 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found description format PCMA for ID 8 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found description format PCMU for ID 0 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found description format GSM for ID 3 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found description format telephone-event for ID 101 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Got unsupported a:fmtp in SDP offer [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: T38 state changed to 0 on channel [Mar 21 11:07:07] VERBOSE[6231] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 21 11:07:07] VERBOSE[6231] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 21 11:07:07] VERBOSE[6231] logger.c: Peer audio RTP is at port 192.168.1.235:8002 [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Checking SIP call limits for device 703 [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Updating call counter for incoming call [Mar 21 11:07:07] VERBOSE[6231] logger.c: Looking for 94006666 in tecnologia (domain 192.168.1.235) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: This channel will not be able to handle video. [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: build_route: Contact hop: [Mar 21 11:07:07] VERBOSE[6231] logger.c: list_route: hop: [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: SIP/703-08230488: New call is still down.... Trying... [Mar 21 11:07:07] VERBOSE[6231] logger.c: <--- Transmitting (no NAT) to 192.168.1.235:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.235:5061;branch=z9hG4bKepvhqzqj;received=192.168.1.235;rport=5061 From: "Miguel" ;tag=ssiea To: Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 276 INVITE User-Agent: Netgate Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 21 11:07:07] DEBUG[6231] devicestate.c: Notification of state change to be queued on device/channel SIP/703-08230488 [Mar 21 11:07:07] DEBUG[6327] pbx.c: Launching 'Macro' [Mar 21 11:07:07] VERBOSE[6327] logger.c: -- Executing [94006666@tecnologia:1] Macro("SIP/703-08230488", "Discar|SIP/spaprueba/4006666") in new stack [Mar 21 11:07:07] DEBUG[6327] pbx.c: Launching 'Set' [Mar 21 11:07:07] VERBOSE[6327] logger.c: -- Executing [s@macro-Discar:1] Set("SIP/703-08230488", "NRADISCAR=SIP/spaprueba/4006666") in new stack [Mar 21 11:07:07] DEBUG[6327] pbx.c: Launching 'Dial' [Mar 21 11:07:07] VERBOSE[6327] logger.c: -- Executing [s@macro-Discar:2] Dial("SIP/703-08230488", "SIP/spaprueba/4006666|40|") in new stack [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Setting NAT on RTP to Off [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: This channel will not be able to handle video. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable STACK-macro-Discar-s-2. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable MACRO_DEPTH. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable NRADISCAR. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable STACK-macro-Discar-s-1. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable ARG1. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable MACRO_PRIORITY. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable MACRO_CONTEXT. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable MACRO_EXTEN. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable STACK-tecnologia-94006666-1. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable SIPCALLID. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable SIPUSERAGENT. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable SIPDOMAIN. [Mar 21 11:07:07] DEBUG[6327] channel.c: Not copying variable SIPURI. [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Outgoing Call for 4006666 [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Updating call counter for outgoing call [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Mar 21 11:07:07] VERBOSE[6327] logger.c: Audio is at 192.168.1.235 port 14958 [Mar 21 11:07:07] VERBOSE[6327] logger.c: Adding codec 0x4 (ulaw) to SDP [Mar 21 11:07:07] VERBOSE[6327] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 0: INVITE sip:4006666@192.168.1.186:5061 SIP/2.0 (45) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK0c6a4a7b;rport (64) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 2: From: "Jose Callero" ;tag=as5245a513 (59) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 3: To: (36) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 4: Contact: (32) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 5: Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 (55) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 7: User-Agent: Netgate (19) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 9: Date: Wed, 21 Mar 2007 14:07:07 GMT (35) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 11: Supported: replaces (19) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 13: Content-Length: 240 (19) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Header 14: (0) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: v=0 (3) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: o=root 6211 6211 IN IP4 192.168.1.235 (37) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: s=session (9) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: c=IN IP4 192.168.1.235 (22) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: t=0 0 (5) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: m=audio 14958 RTP/AVP 0 101 (27) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: a=ptime:20 (10) [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Line: a=sendrecv (10) [Mar 21 11:07:07] VERBOSE[6327] logger.c: Reliably Transmitting (no NAT) to 192.168.1.186:5061: INVITE sip:4006666@192.168.1.186:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK0c6a4a7b;rport From: "Jose Callero" ;tag=as5245a513 To: Contact: Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 CSeq: 102 INVITE User-Agent: Netgate Max-Forwards: 70 Date: Wed, 21 Mar 2007 14:07:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 6211 6211 IN IP4 192.168.1.235 s=session c=IN IP4 192.168.1.235 t=0 0 m=audio 14958 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #181 [Mar 21 11:07:07] VERBOSE[6327] logger.c: -- Called spaprueba/4006666 [Mar 21 11:07:07] VERBOSE[6231] logger.c: <--- SIP read from 192.168.1.186:5061 ---> SIP/2.0 100 Trying To: From: "Jose Callero" ;tag=as5245a513 Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK0c6a4a7b Server: Linksys/SPA3102-3.3.6(GW) Content-Length: 0 <-------------> [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 1: To: (36) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 2: From: "Jose Callero" ;tag=as5245a513 (59) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 3: Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 (55) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 4: CSeq: 102 INVITE (16) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK0c6a4a7b (58) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 6: Server: Linksys/SPA3102-3.3.6(GW) (33) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 7: Content-Length: 0 (17) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 8: (0) [Mar 21 11:07:07] VERBOSE[6231] logger.c: --- (8 headers 0 lines) --- [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: *** SIP TIMER: Cancelling retransmission #181 - INVITE (got response) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235' Request 102: Found [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: SIP response 100 to standard invite [Mar 21 11:07:07] DEBUG[6214] devicestate.c: No provider found, checking channel drivers for SIP - 703 [Mar 21 11:07:07] DEBUG[6214] chan_sip.c: Checking device state for peer 703 [Mar 21 11:07:07] DEBUG[6214] devicestate.c: Changing state for SIP/703 - state 1 (Not in use) [Mar 21 11:07:07] DEBUG[6328] app_queue.c: Device 'SIP/703' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 11:07:07] VERBOSE[6231] logger.c: <--- SIP read from 192.168.1.186:5061 ---> SIP/2.0 180 Ringing To: ;tag=39ff1524212a9a8di1 From: "Jose Callero" ;tag=as5245a513 Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK0c6a4a7b Server: Linksys/SPA3102-3.3.6(GW) Remote-Party-ID: spaprueba ;screen=yes;party=called Content-Length: 0 <-------------> [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 1: To: ;tag=39ff1524212a9a8di1 (59) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 2: From: "Jose Callero" ;tag=as5245a513 (59) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 3: Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 (55) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 4: CSeq: 102 INVITE (16) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK0c6a4a7b (58) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 6: Server: Linksys/SPA3102-3.3.6(GW) (33) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 7: Remote-Party-ID: spaprueba ;screen=yes;party=called (80) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 8: Content-Length: 0 (17) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 9: (0) [Mar 21 11:07:07] VERBOSE[6231] logger.c: --- (9 headers 0 lines) --- [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235' Request 102: Found [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: SIP response 180 to standard invite [Mar 21 11:07:07] DEBUG[6231] devicestate.c: Notification of state change to be queued on device/channel SIP/spaprueba-0822de18 [Mar 21 11:07:07] VERBOSE[6327] logger.c: -- SIP/spaprueba-0822de18 is ringing [Mar 21 11:07:07] VERBOSE[6231] logger.c: <--- SIP read from 192.168.1.186:5061 ---> SIP/2.0 200 OK To: ;tag=39ff1524212a9a8di1 From: "Jose Callero" ;tag=as5245a513 Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK0c6a4a7b Contact: spaprueba Server: Linksys/SPA3102-3.3.6(GW) Remote-Party-ID: spaprueba ;screen=yes;party=called Content-Length: 251 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 6882 6882 IN IP4 192.168.1.186 s=- c=IN IP4 192.168.1.186 t=0 0 m=audio 16420 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 1: To: ;tag=39ff1524212a9a8di1 (59) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 2: From: "Jose Callero" ;tag=as5245a513 (59) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 3: Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 (55) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 4: CSeq: 102 INVITE (16) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK0c6a4a7b (58) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 6: Contact: spaprueba (51) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 7: Server: Linksys/SPA3102-3.3.6(GW) (33) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 8: Remote-Party-ID: spaprueba ;screen=yes;party=called (80) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 9: Content-Length: 251 (19) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 10: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 11: Supported: x-sipura (19) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Header 13: (0) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: v=0 (3) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: o=- 6882 6882 IN IP4 192.168.1.186 (34) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: s=- (3) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: c=IN IP4 192.168.1.186 (22) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: t=0 0 (5) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: m=audio 16420 RTP/AVP 0 100 101 (31) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:100 NSE/8000 (21) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=fmtp:100 192-193 (18) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=ptime:30 (10) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Line: a=sendrecv (10) [Mar 21 11:07:07] VERBOSE[6231] logger.c: --- (13 headers 13 lines) --- [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Acked pending invite 102 [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Stopping retransmission on '0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235' of Request 102: Match Not Found [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: SIP response 200 to standard invite [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found RTP audio format 0 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found RTP audio format 100 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found RTP audio format 101 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Peer audio RTP is at port 192.168.1.186:16420 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found description format PCMU for ID 0 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found description format NSE for ID 100 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Got unsupported a:fmtp in SDP offer [Mar 21 11:07:07] VERBOSE[6231] logger.c: Found description format telephone-event for ID 101 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Got unsupported a:fmtp in SDP offer [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: T38 state changed to 0 on channel SIP/spaprueba-0822de18 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Mar 21 11:07:07] VERBOSE[6231] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Mar 21 11:07:07] VERBOSE[6231] logger.c: Peer audio RTP is at port 192.168.1.186:16420 [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: We have an owner, now see if we need to change this call [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Updating call counter for outgoing call [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: build_route: Contact hop: spaprueba [Mar 21 11:07:07] VERBOSE[6231] logger.c: list_route: hop: [Mar 21 11:07:07] DEBUG[6231] chan_sip.c: Strict routing enforced for session 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 [Mar 21 11:07:07] VERBOSE[6231] logger.c: set_destination: Parsing for address/port to send to [Mar 21 11:07:07] VERBOSE[6231] logger.c: set_destination: set destination to 192.168.1.186, port 5061 [Mar 21 11:07:07] VERBOSE[6231] logger.c: Transmitting (no NAT) to 192.168.1.186:5061: ACK sip:4006666@192.168.1.186:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK745dcd29;rport From: "Jose Callero" ;tag=as5245a513 To: ;tag=39ff1524212a9a8di1 Contact: Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 CSeq: 102 ACK User-Agent: Netgate Max-Forwards: 70 Content-Length: 0 --- [Mar 21 11:07:07] VERBOSE[6327] logger.c: <--- Transmitting (no NAT) to 192.168.1.235:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.235:5061;branch=z9hG4bKepvhqzqj;received=192.168.1.235;rport=5061 From: "Miguel" ;tag=ssiea To: ;tag=as6de36e35 Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 276 INVITE User-Agent: Netgate Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 21 11:07:07] VERBOSE[6327] logger.c: -- Call on SIP/spaprueba-0822de18 left from hold [Mar 21 11:07:07] DEBUG[6327] devicestate.c: Notification of state change to be queued on device/channel SIP/spaprueba-0822de18 [Mar 21 11:07:07] VERBOSE[6327] logger.c: -- SIP/spaprueba-0822de18 answered SIP/703-08230488 [Mar 21 11:07:07] DEBUG[6327] devicestate.c: Notification of state change to be queued on device/channel SIP/703-08230488 [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: SIP answering channel: SIP/703-08230488 [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Setting framing from config on incoming call [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Mar 21 11:07:07] VERBOSE[6327] logger.c: Audio is at 192.168.1.235 port 15960 [Mar 21 11:07:07] VERBOSE[6327] logger.c: Adding codec 0x4 (ulaw) to SDP [Mar 21 11:07:07] VERBOSE[6327] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: -- Done with adding codecs to SDP [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Mar 21 11:07:07] VERBOSE[6327] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.1.235:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.235:5061;branch=z9hG4bKepvhqzqj;received=192.168.1.235;rport=5061 From: "Miguel" ;tag=ssiea To: ;tag=as6de36e35 Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 276 INVITE User-Agent: Netgate Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 6211 6211 IN IP4 192.168.1.235 s=session c=IN IP4 192.168.1.235 t=0 0 m=audio 15960 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Mar 21 11:07:07] DEBUG[6327] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #183 [Mar 21 11:07:07] VERBOSE[6327] logger.c: -- Packet2Packet bridging SIP/703-08230488 and SIP/spaprueba-0822de18 [Mar 21 11:07:07] DEBUG[6214] devicestate.c: No provider found, checking channel drivers for SIP - spaprueba [Mar 21 11:07:07] DEBUG[6214] chan_sip.c: Checking device state for peer spaprueba [Mar 21 11:07:07] DEBUG[6214] devicestate.c: Changing state for SIP/spaprueba - state 1 (Not in use) [Mar 21 11:07:07] DEBUG[6214] devicestate.c: No provider found, checking channel drivers for SIP - spaprueba [Mar 21 11:07:07] DEBUG[6214] chan_sip.c: Checking device state for peer spaprueba [Mar 21 11:07:07] DEBUG[6214] devicestate.c: Changing state for SIP/spaprueba - state 1 (Not in use) [Mar 21 11:07:07] DEBUG[6214] devicestate.c: No provider found, checking channel drivers for SIP - 703 [Mar 21 11:07:07] DEBUG[6214] chan_sip.c: Checking device state for peer 703 [Mar 21 11:07:07] DEBUG[6214] devicestate.c: Changing state for SIP/703 - state 1 (Not in use) [Mar 21 11:07:08] DEBUG[6330] app_queue.c: Device 'SIP/spaprueba' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 11:07:08] DEBUG[6331] app_queue.c: Device 'SIP/spaprueba' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 11:07:08] DEBUG[6332] app_queue.c: Device 'SIP/703' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 11:07:08] VERBOSE[6231] logger.c: <--- SIP read from 192.168.1.235:5061 ---> ACK sip:94006666@192.168.1.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKuanukfch Max-Forwards: 70 Proxy-Authorization: Digest username="703",realm="asterisk",nonce="00559a27",uri="sip:94006666@192.168.1.235",response="d749b296e592f2e87020fb9df7bea852",algorithm=MD5 To: ;tag=as6de36e35 From: "Miguel" ;tag=ssiea Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 276 ACK User-Agent: Twinkle/0.9 Content-Length: 0 <-------------> [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 0: ACK sip:94006666@192.168.1.235 SIP/2.0 (38) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKuanukfch (64) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 3: Proxy-Authorization: Digest username="703",realm="asterisk",nonce="00559a27",uri="sip:94006666@192.168.1.235",response="d749b296e592f2e87020fb9df7bea852",algorithm=MD5 (167) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 4: To: ;tag=as6de36e35 (47) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 5: From: "Miguel" ;tag=ssiea (48) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 6: Call-ID: ttcjtjstxvndzeg@192.168.1.235 (38) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 7: CSeq: 276 ACK (13) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 8: User-Agent: Twinkle/0.9 (23) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 9: Content-Length: 0 (17) [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Header 10: (0) [Mar 21 11:07:08] VERBOSE[6231] logger.c: --- (10 headers 0 lines) --- [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #183 [Mar 21 11:07:08] DEBUG[6231] chan_sip.c: Stopping retransmission on 'ttcjtjstxvndzeg@192.168.1.235' of Response 276: Match Not Found [Mar 21 11:07:08] DEBUG[6327] rtp.c: Got RTCP report of 44 bytes [Mar 21 11:07:14] DEBUG[6327] rtp.c: Got RTCP report of 88 bytes [Mar 21 11:07:17] VERBOSE[6231] logger.c: <--- SIP read from 192.168.1.235:5061 ---> BYE sip:94006666@192.168.1.235 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKimhyiexv Max-Forwards: 70 To: ;tag=as6de36e35 From: "Miguel" ;tag=ssiea Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 277 BYE User-Agent: Twinkle/0.9 Content-Length: 0 <-------------> [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 0: BYE sip:94006666@192.168.1.235 SIP/2.0 (38) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.235:5061;rport;branch=z9hG4bKimhyiexv (64) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 3: To: ;tag=as6de36e35 (47) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 4: From: "Miguel" ;tag=ssiea (48) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 5: Call-ID: ttcjtjstxvndzeg@192.168.1.235 (38) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 6: CSeq: 277 BYE (13) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 7: User-Agent: Twinkle/0.9 (23) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 8: Content-Length: 0 (17) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 9: (0) [Mar 21 11:07:17] VERBOSE[6231] logger.c: --- (9 headers 0 lines) --- [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Mar 21 11:07:17] VERBOSE[6231] logger.c: Sending to 192.168.1.235 : 5061 (NAT) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Setting SIP_ALREADYGONE on dialog ttcjtjstxvndzeg@192.168.1.235 [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Received bye, issuing owner hangup [Mar 21 11:07:17] VERBOSE[6231] logger.c: <--- Transmitting (NAT) to 192.168.1.235:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.235:5061;branch=z9hG4bKimhyiexv;received=192.168.1.235;rport=5061 From: "Miguel" ;tag=ssiea To: ;tag=as6de36e35 Call-ID: ttcjtjstxvndzeg@192.168.1.235 CSeq: 277 BYE User-Agent: Netgate Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 21 11:07:17] DEBUG[6327] rtp.c: Oooh, got a hangup [Mar 21 11:07:17] DEBUG[6327] channel.c: Returning from native bridge, channels: SIP/703-08230488, SIP/spaprueba-0822de18 [Mar 21 11:07:17] DEBUG[6327] channel.c: Hanging up channel 'SIP/spaprueba-0822de18' [Mar 21 11:07:17] DEBUG[6327] chan_sip.c: Hangup call SIP/spaprueba-0822de18, SIP callid 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235) [Mar 21 11:07:17] VERBOSE[6327] logger.c: Scheduling destruction of SIP dialog '0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235' in 6400 ms (Method: INVITE) [Mar 21 11:07:17] DEBUG[6327] chan_sip.c: Strict routing enforced for session 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 [Mar 21 11:07:17] VERBOSE[6327] logger.c: set_destination: Parsing for address/port to send to [Mar 21 11:07:17] VERBOSE[6327] logger.c: set_destination: set destination to 192.168.1.186, port 5061 [Mar 21 11:07:17] VERBOSE[6327] logger.c: Reliably Transmitting (no NAT) to 192.168.1.186:5061: BYE sip:4006666@192.168.1.186:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK242b00a2;rport From: "Jose Callero" ;tag=as5245a513 To: ;tag=39ff1524212a9a8di1 Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 CSeq: 103 BYE User-Agent: Netgate Max-Forwards: 70 Content-Length: 0 --- [Mar 21 11:07:17] DEBUG[6327] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #185 [Mar 21 11:07:17] DEBUG[6327] devicestate.c: Notification of state change to be queued on device/channel SIP/spaprueba-0822de18 [Mar 21 11:07:17] DEBUG[6327] rtp.c: Channel '' has no RTP, not doing anything [Mar 21 11:07:17] DEBUG[6327] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Mar 21 11:07:17] DEBUG[6327] app_macro.c: Spawn extension (macro-Discar,s,2) exited non-zero on 'SIP/703-08230488' in macro 'Discar' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Spawn extension (macro-Discar,s,2) exited non-zero on 'SIP/703-08230488' [Mar 21 11:07:17] VERBOSE[6327] logger.c: == Spawn extension (macro-Discar, s, 2) exited non-zero on 'SIP/703-08230488' [Mar 21 11:07:17] VERBOSE[6231] logger.c: <--- SIP read from 192.168.1.186:5061 ---> SIP/2.0 200 OK To: ;tag=39ff1524212a9a8di1 From: "Jose Callero" ;tag=as5245a513 Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 CSeq: 103 BYE Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK242b00a2 Server: Linksys/SPA3102-3.3.6(GW) Content-Length: 0 <-------------> [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 1: To: ;tag=39ff1524212a9a8di1 (59) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 2: From: "Jose Callero" ;tag=as5245a513 (59) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 3: Call-ID: 0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235 (55) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 4: CSeq: 103 BYE (13) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.235:5060;branch=z9hG4bK242b00a2 (58) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 6: Server: Linksys/SPA3102-3.3.6(GW) (33) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 7: Content-Length: 0 (17) [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Header 8: (0) [Mar 21 11:07:17] VERBOSE[6231] logger.c: --- (8 headers 0 lines) --- [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #185 [Mar 21 11:07:17] DEBUG[6231] chan_sip.c: Stopping retransmission on '0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235' of Request 103: Match Not Found [Mar 21 11:07:17] VERBOSE[6231] logger.c: Really destroying SIP dialog '0f5e384c43d9e26f2c9a537b4c140669@192.168.1.235' Method: INVITE [Mar 21 11:07:17] DEBUG[6214] devicestate.c: No provider found, checking channel drivers for SIP - spaprueba [Mar 21 11:07:17] DEBUG[6214] chan_sip.c: Checking device state for peer spaprueba [Mar 21 11:07:17] DEBUG[6214] devicestate.c: Changing state for SIP/spaprueba - state 1 (Not in use) [Mar 21 11:07:17] DEBUG[6336] app_queue.c: Device 'SIP/spaprueba' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 11:07:17] DEBUG[6327] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record. [Mar 21 11:07:17] DEBUG[6327] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield) VALUES ('2007-03-21 11:07:07','\"Jose Callero\" <703>','703','94006666','tecnologia', 'SIP/703-08230488','SIP/spaprueba-0822de18','Dial','SIP/spaprueba/4006666|40|',10,10,'ANSWERED',3,'','') [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '"Jose Callero" <703>' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '703' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '94006666' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is 'tecnologia' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is 'SIP/703-08230488' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is 'SIP/spaprueba-0822de18' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is 'Dial' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is 'SIP/spaprueba/4006666|40|' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '2007-03-21 11:07:07' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '2007-03-21 11:07:07' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '2007-03-21 11:07:17' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '10' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '10' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is 'ANSWERED' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is 'DOCUMENTATION' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '1174486027.6' [Mar 21 11:07:17] DEBUG[6327] pbx.c: Function result is '' [Mar 21 11:07:17] DEBUG[6327] channel.c: Hanging up channel 'SIP/703-08230488' [Mar 21 11:07:17] DEBUG[6327] chan_sip.c: Hangup call SIP/703-08230488, SIP callid ttcjtjstxvndzeg@192.168.1.235) [Mar 21 11:07:17] DEBUG[6327] devicestate.c: Notification of state change to be queued on device/channel SIP/703-08230488 [Mar 21 11:07:17] DEBUG[6214] devicestate.c: No provider found, checking channel drivers for SIP - 703 [Mar 21 11:07:17] DEBUG[6214] chan_sip.c: Checking device state for peer 703 [Mar 21 11:07:17] DEBUG[6214] devicestate.c: Changing state for SIP/703 - state 1 (Not in use) [Mar 21 11:07:17] DEBUG[6337] app_queue.c: Device 'SIP/703' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 21 11:07:18] VERBOSE[6231] logger.c: Really destroying SIP dialog 'ttcjtjstxvndzeg@192.168.1.235' Method: BYE [Mar 21 11:07:21] VERBOSE[6237] logger.c: Beginning asterisk shutdown.... [Mar 21 11:07:21] VERBOSE[6237] logger.c: Executing last minute cleanups [Mar 21 11:07:21] VERBOSE[6237] logger.c: == Destroying musiconhold processes [Mar 21 11:07:21] VERBOSE[6237] logger.c: Asterisk cleanly ending (0). [Mar 21 11:07:21] DEBUG[6237] asterisk.c: Asterisk ending (0).