[Apr 3 03:51:02] VERBOSE[20549] logger.c: <--- SIP read from 111.111.111.111:5060 ---> INVITE sip:10123456789@asterisk SIP/2.0 Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK4947a09f;rport From: "0000000000" ;tag=as0d87f493 To: Contact: Call-ID: 053893d774d6282f768892985e462f4d@111.111.111.111 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 03 Apr 2007 07:51:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 287 v=0 o=root 2808 2808 IN IP4 111.111.111.111 s=session c=IN IP4 111.111.111.111 t=0 0 m=audio 14148 RTP/AVP 8 0 3 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 0: INVITE sip:10123456789@asterisk SIP/2.0 (51) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 1: Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK4947a09f;rport (63) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 2: From: "0000000000" ;tag=as0d87f493 (63) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 3: To: (42) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 4: Contact: (38) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 5: Call-ID: 053893d774d6282f768892985e462f4d@111.111.111.111 (54) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 9: Date: Tue, 03 Apr 2007 07:51:02 GMT (35) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 11: Content-Type: application/sdp (29) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 12: Content-Length: 287 (19) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 13: (0) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: v=0 (3) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: o=root 2808 2808 IN IP4 111.111.111.111 (36) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: s=session (9) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: c=IN IP4 111.111.111.111 (21) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: t=0 0 (5) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: m=audio 14148 RTP/AVP 8 0 3 97 101 (34) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: a=rtpmap:97 iLBC/8000 (21) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 3 03:51:02] VERBOSE[20549] logger.c: --- (13 headers 13 lines) --- [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Setting NAT on RTP to Off [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Setting NAT on VRTP to Off [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Setting NAT on UDPTL to Off [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Allocating new SIP dialog for 053893d774d6282f768892985e462f4d@111.111.111.111 - INVITE (With RTP) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Apr 3 03:51:02] VERBOSE[20549] logger.c: Sending to 111.111.111.111 : 5060 (NAT) [Apr 3 03:51:02] VERBOSE[20549] logger.c: Using INVITE request as basis request - 053893d774d6282f768892985e462f4d@111.111.111.111 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found peer 'home' [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Setting NAT on RTP to Off [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Setting NAT on VRTP to Off [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Setting NAT on UDPTL to Off [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found RTP audio format 8 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found RTP audio format 0 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found RTP audio format 3 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found RTP audio format 97 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found RTP audio format 101 [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Peer doesn't provide T.38 UDPTL [Apr 3 03:51:02] VERBOSE[20549] logger.c: Peer audio RTP is at port 111.111.111.111:14148 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found description format PCMA for ID 8 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found description format PCMU for ID 0 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found description format GSM for ID 3 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found description format iLBC for ID 97 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Found description format telephone-event for ID 101 [Apr 3 03:51:02] VERBOSE[20549] logger.c: Got unsupported a:fmtp in SDP offer [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: T38 state changed to 0 on channel [Apr 3 03:51:02] VERBOSE[20549] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Apr 3 03:51:02] VERBOSE[20549] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Apr 3 03:51:02] VERBOSE[20549] logger.c: Peer audio RTP is at port 111.111.111.111:14148 [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Checking SIP call limits for device [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Updating call counter for incoming call [Apr 3 03:51:02] VERBOSE[20549] logger.c: Looking for 10123456789 in home (domain asterisk) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: *** Our native formats are 0x2 (gsm) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x2 (gsm) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: This channel will not be able to handle video. [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: build_route: Contact hop: [Apr 3 03:51:02] VERBOSE[20549] logger.c: list_route: hop: [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: SIP/111.111.111.111-09d218f0: New call is still down.... Trying... [Apr 3 03:51:02] VERBOSE[20549] logger.c: <--- Transmitting (no NAT) to 111.111.111.111:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK4947a09f;received=111.111.111.111;rport=5060 From: "0000000000" ;tag=as0d87f493 To: Call-ID: 053893d774d6282f768892985e462f4d@111.111.111.111 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Apr 3 03:51:02] DEBUG[20549] devicestate.c: Notification of state change to be queued on device/channel SIP/111.111.111.111-09d218f0 [Apr 3 03:51:02] DEBUG[20543] devicestate.c: No provider found, checking channel drivers for SIP - 111.111.111.111 [Apr 3 03:51:02] DEBUG[20543] chan_sip.c: Checking device state for peer 111.111.111.111 [Apr 3 03:51:02] DEBUG[20557] pbx.c: Launching 'Dial' [Apr 3 03:51:02] VERBOSE[20557] logger.c: -- Executing [10123456789@home:1] Dial("SIP/111.111.111.111-09d218f0", "SIP/provider/10123456789") in new stack [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Asked to create a SIP channel with formats: 0x2 (gsm) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Our T38 capability (3856) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Setting NAT on RTP to Off [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Setting NAT on UDPTL to Off [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: *** Our native formats are 0x100 (g729) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: *** Our capabilities are 0x104 (ulaw|g729) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: *** Our preferred formats from the incoming channel are 0x2 (gsm) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: This channel will not be able to handle video. [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Early remote bridge setting SIP '6331a7393f9711700180ce7e07c43b51@333.333.333.333' - Sending media to 111.111.111.111 [Apr 3 03:51:02] DEBUG[20557] rtp.c: Seeded SDP of 'SIP/provider-09d35848' with that of 'SIP/111.111.111.111-09d218f0' [Apr 3 03:51:02] DEBUG[20557] channel.c: Not copying variable STACK-home-10123456789-1. [Apr 3 03:51:02] DEBUG[20557] channel.c: Not copying variable SIPCALLID. [Apr 3 03:51:02] DEBUG[20557] channel.c: Not copying variable SIPUSERAGENT. [Apr 3 03:51:02] DEBUG[20557] channel.c: Not copying variable SIPDOMAIN. [Apr 3 03:51:02] DEBUG[20557] channel.c: Not copying variable SIPURI. [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Outgoing Call for 10123456789 [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Updating call counter for outgoing call [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Our T38 capability (3856), joint T38 capability (3856) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: ** Our capability: 0x104 (ulaw|g729) Video flag: False [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: ** Our prefcodec: 0x2 (gsm) [Apr 3 03:51:02] VERBOSE[20557] logger.c: Audio is at 333.333.333.333 port 11468 [Apr 3 03:51:02] VERBOSE[20557] logger.c: Adding codec 0x100 (g729) to SDP [Apr 3 03:51:02] VERBOSE[20557] logger.c: Adding codec 0x4 (ulaw) to SDP [Apr 3 03:51:02] VERBOSE[20557] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: -- Done with adding codecs to SDP [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Done building SDP. Settling with this capability: 0x104 (ulaw|g729) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 0: INVITE sip:10123456789@222.222.222.222:5061 SIP/2.0 (50) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 1: Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport (64) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 2: From: "0000000000" ;tag=as095627b6 (64) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 3: To: (41) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 4: Contact: (39) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 5: Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 (55) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 7: User-Agent: Asterisk (18) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 9: Remote-Party-ID: "0000000000" ;privacy=off;screen=no (82) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 10: Date: Tue, 03 Apr 2007 07:51:02 GMT (35) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 12: Supported: replaces (19) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 14: Content-Length: 287 (19) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Header 15: (0) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: v=0 (3) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: o=root 20536 20536 IN IP4 111.111.111.111 (38) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: s=session (9) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: c=IN IP4 111.111.111.111 (21) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: t=0 0 (5) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: m=audio 14148 RTP/AVP 18 0 101 (30) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: a=ptime:20 (10) [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: Line: a=sendrecv (10) [Apr 3 03:51:02] VERBOSE[20557] logger.c: Reliably Transmitting (no NAT) to 222.222.222.222:5061: INVITE sip:10123456789@222.222.222.222:5061 SIP/2.0 Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport From: "0000000000" ;tag=as095627b6 To: Contact: Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 70 Remote-Party-ID: "0000000000" ;privacy=off;screen=no Date: Tue, 03 Apr 2007 07:51:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 20536 20536 IN IP4 111.111.111.111 s=session c=IN IP4 111.111.111.111 t=0 0 m=audio 14148 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Apr 3 03:51:02] DEBUG[20557] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #270 [Apr 3 03:51:02] VERBOSE[20557] logger.c: -- Called provider/10123456789 [Apr 3 03:51:02] DEBUG[20557] channel.c: Set channel SIP/provider-09d35848 to read format slin [Apr 3 03:51:02] DEBUG[20557] channel.c: Set channel SIP/111.111.111.111-09d218f0 to write format slin [Apr 3 03:51:02] DEBUG[20557] channel.c: Set channel SIP/111.111.111.111-09d218f0 to read format slin [Apr 3 03:51:02] DEBUG[20557] channel.c: Set channel SIP/provider-09d35848 to write format slin [Apr 3 03:51:02] DEBUG[20543] devicestate.c: Changing state for SIP/111.111.111.111 - state 2 (In use) [Apr 3 03:51:02] VERBOSE[20549] logger.c: <--- SIP read from 222.222.222.222:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport=5060;received=333.333.333.333 From: "0000000000" ;tag=as095627b6 To: Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 CSeq: 102 INVITE Server: MERA MSIP v.3.0 Content-Length: 0 <-------------> [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 1: Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport=5060;received=333.333.333.333 (92) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 2: From: "0000000000" ;tag=as095627b6 (64) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 3: To: (41) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 4: Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 (55) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 6: Server: MERA MSIP v.3.0 (23) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 7: Content-Length: 0 (17) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: Header 8: (0) [Apr 3 03:51:02] VERBOSE[20549] logger.c: --- (8 headers 0 lines) --- [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: *** SIP TIMER: Cancelling retransmission #270 - INVITE (got response) [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6331a7393f9711700180ce7e07c43b51@333.333.333.333' Request 102: Found [Apr 3 03:51:02] DEBUG[20549] chan_sip.c: SIP response 100 to standard invite [Apr 3 03:51:03] DEBUG[20558] manager.c: Manager received command 'Login' [Apr 3 03:51:03] VERBOSE[20558] logger.c: == Parsing '/etc/asterisk/manager.conf': [Apr 3 03:51:03] DEBUG[20558] config.c: Parsing /etc/asterisk/manager.conf [Apr 3 03:51:03] VERBOSE[20558] logger.c: Found [Apr 3 03:51:03] DEBUG[20558] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer [Apr 3 03:51:03] DEBUG[20558] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer [Apr 3 03:51:03] DEBUG[20558] acl.c: ##### Testing 127.0.0.1 with 0.0.0.0 [Apr 3 03:51:03] DEBUG[20558] acl.c: ##### Testing 127.0.0.1 with 127.0.0.0 [Apr 3 03:51:03] DEBUG[20558] manager.c: Manager received command 'Command' [Apr 3 03:51:03] DEBUG[20558] manager.c: Manager received command 'Logoff' [Apr 3 03:51:04] VERBOSE[20549] logger.c: <--- SIP read from 222.222.222.222:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport=5060;received=333.333.333.333 From: "0000000000" ;tag=as095627b6 To: ;tag=6627200-269616998-352321664-658962373 Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 CSeq: 102 INVITE Contact: Server: MERA MSIP v.3.0 Content-Length: 0 <-------------> [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 1: Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport=5060;received=333.333.333.333 (92) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 2: From: "0000000000" ;tag=as095627b6 (64) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 3: To: ;tag=6627200-269616998-352321664-658962373 (83) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 4: Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 (55) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 6: Contact: (46) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 7: Server: MERA MSIP v.3.0 (23) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 8: Content-Length: 0 (17) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 9: (0) [Apr 3 03:51:04] VERBOSE[20549] logger.c: --- (9 headers 0 lines) --- [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6331a7393f9711700180ce7e07c43b51@333.333.333.333' Request 102: Found [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: SIP response 180 to standard invite [Apr 3 03:51:04] DEBUG[20549] devicestate.c: Notification of state change to be queued on device/channel SIP/provider-09d35848 [Apr 3 03:51:04] DEBUG[20543] devicestate.c: No provider found, checking channel drivers for SIP - provider [Apr 3 03:51:04] DEBUG[20543] chan_sip.c: Checking device state for peer provider [Apr 3 03:51:04] DEBUG[20543] devicestate.c: Changing state for SIP/provider - state 1 (Not in use) [Apr 3 03:51:04] VERBOSE[20557] logger.c: -- SIP/provider-09d35848 is ringing [Apr 3 03:51:04] DEBUG[20557] chan_sip.c: Early remote bridge setting SIP '053893d774d6282f768892985e462f4d@111.111.111.111' - Sending media to 333.333.333.333 [Apr 3 03:51:04] DEBUG[20557] rtp.c: Setting early bridge SDP of 'SIP/111.111.111.111-09d218f0' with that of 'SIP/provider-09d35848' [Apr 3 03:51:04] VERBOSE[20557] logger.c: <--- Transmitting (no NAT) to 111.111.111.111:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK4947a09f;received=111.111.111.111;rport=5060 From: "0000000000" ;tag=as0d87f493 To: ;tag=as3f469b61 Call-ID: 053893d774d6282f768892985e462f4d@111.111.111.111 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Apr 3 03:51:04] DEBUG[20557] channel.c: Driver for channel 'SIP/111.111.111.111-09d218f0' does not support indication 3, emulating it [Apr 3 03:51:04] DEBUG[20557] channel.c: Prodding channel 'SIP/111.111.111.111-09d218f0' [Apr 3 03:51:04] DEBUG[20557] chan_sip.c: Setting framing from config on incoming call [Apr 3 03:51:04] DEBUG[20557] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Apr 3 03:51:04] DEBUG[20557] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Apr 3 03:51:04] VERBOSE[20557] logger.c: Audio is at 333.333.333.333 port 17466 [Apr 3 03:51:04] VERBOSE[20557] logger.c: Adding codec 0x2 (gsm) to SDP [Apr 3 03:51:04] VERBOSE[20557] logger.c: Adding codec 0x4 (ulaw) to SDP [Apr 3 03:51:04] VERBOSE[20557] logger.c: Adding codec 0x8 (alaw) to SDP [Apr 3 03:51:04] VERBOSE[20557] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 3 03:51:04] DEBUG[20557] chan_sip.c: -- Done with adding codecs to SDP [Apr 3 03:51:04] DEBUG[20557] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Apr 3 03:51:04] VERBOSE[20557] logger.c: <--- Transmitting (no NAT) to 111.111.111.111:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK4947a09f;received=111.111.111.111;rport=5060 From: "0000000000" ;tag=as0d87f493 To: ;tag=as3f469b61 Call-ID: 053893d774d6282f768892985e462f4d@111.111.111.111 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 20536 20536 IN IP4 333.333.333.333 s=session c=IN IP4 333.333.333.333 t=0 0 m=audio 17466 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Apr 3 03:51:04] DEBUG[20557] channel.c: Scheduling timer at 160 sample intervals [Apr 3 03:51:04] DEBUG[20557] rtp.c: Ooh, format changed from unknown to gsm [Apr 3 03:51:04] DEBUG[20557] rtp.c: Created smoother: format: 2 ms: 20 len: 33 [Apr 3 03:51:04] DEBUG[20557] channel.c: Generator got voice, switching to phase locked mode [Apr 3 03:51:04] DEBUG[20557] channel.c: Scheduling timer at 0 sample intervals [Apr 3 03:51:04] VERBOSE[20549] logger.c: <--- SIP read from 222.222.222.222:5061 ---> SIP/2.0 183 Progress Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport=5060;received=333.333.333.333 From: "0000000000" ;tag=as095627b6 To: ;tag=6627200-269616998-352321664-658962373 Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 CSeq: 102 INVITE Contact: Server: MERA MSIP v.3.0 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1175586664 1175586664 IN IP4 222.222.222.222 s=- c=IN IP4 222.222.222.222 t=0 0 m=audio 22480 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 0: SIP/2.0 183 Progress (20) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 1: Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport=5060;received=333.333.333.333 (92) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 2: From: "0000000000" ;tag=as095627b6 (64) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 3: To: ;tag=6627200-269616998-352321664-658962373 (83) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 4: Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 (55) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 6: Contact: (46) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 7: Server: MERA MSIP v.3.0 (23) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 8: Content-Type: application/sdp (29) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 9: Content-Length: 241 (21) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 10: (0) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: v=0 (3) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: o=- 1175586664 1175586664 IN IP4 222.222.222.222 (47) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: s=- (3) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: c=IN IP4 222.222.222.222 (23) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: t=0 0 (5) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: m=audio 22480 RTP/AVP 18 0 101 (30) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Apr 3 03:51:04] VERBOSE[20549] logger.c: --- (10 headers 11 lines) --- [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6331a7393f9711700180ce7e07c43b51@333.333.333.333' Request 102: Found [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: SIP response 183 to standard invite [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found RTP audio format 18 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found RTP audio format 0 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found RTP audio format 101 [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Peer doesn't provide T.38 UDPTL [Apr 3 03:51:04] VERBOSE[20549] logger.c: Peer audio RTP is at port 222.222.222.222:22480 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found description format G729 for ID 18 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Got unsupported a:fmtp in SDP offer [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found description format PCMU for ID 0 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found description format telephone-event for ID 101 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Got unsupported a:fmtp in SDP offer [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: T38 state changed to 0 on channel SIP/provider-09d35848 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) [Apr 3 03:51:04] VERBOSE[20549] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Apr 3 03:51:04] VERBOSE[20549] logger.c: Peer audio RTP is at port 222.222.222.222:22480 [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: We're settling with these formats: 0x104 (ulaw|g729) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: We have an owner, now see if we need to change this call [Apr 3 03:51:04] VERBOSE[20557] logger.c: -- Call on SIP/provider-09d35848 left from hold [Apr 3 03:51:04] DEBUG[20557] channel.c: Scheduling timer at 0 sample intervals [Apr 3 03:51:04] VERBOSE[20557] logger.c: -- SIP/provider-09d35848 is making progress passing it to SIP/111.111.111.111-09d218f0 [Apr 3 03:51:04] DEBUG[20557] chan_sip.c: Early remote bridge setting SIP '053893d774d6282f768892985e462f4d@111.111.111.111' - Sending media to 222.222.222.222 [Apr 3 03:51:04] DEBUG[20557] rtp.c: Setting early bridge SDP of 'SIP/111.111.111.111-09d218f0' with that of 'SIP/provider-09d35848' [Apr 3 03:51:04] DEBUG[20557] rtp.c: Ooh, format changed from unknown to g729 [Apr 3 03:51:04] DEBUG[20557] rtp.c: Created smoother: format: 256 ms: 20 len: 20 [Apr 3 03:51:04] VERBOSE[20549] logger.c: <--- SIP read from 222.222.222.222:5061 ---> SIP/2.0 183 Progress Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport=5060;received=333.333.333.333 From: "0000000000" ;tag=as095627b6 To: ;tag=6627200-269616998-352321664-658962373 Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 CSeq: 102 INVITE Contact: Server: MERA MSIP v.3.0 Content-Type: application/sdp Content-Length: 241 v=0 o=- 1175586664 1175586664 IN IP4 222.222.222.222 s=- c=IN IP4 222.222.222.222 t=0 0 m=audio 22480 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 0: SIP/2.0 183 Progress (20) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 1: Via: SIP/2.0/UDP 333.333.333.333:5060;branch=z9hG4bK7871899c;rport=5060;received=333.333.333.333 (92) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 2: From: "0000000000" ;tag=as095627b6 (64) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 3: To: ;tag=6627200-269616998-352321664-658962373 (83) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 4: Call-ID: 6331a7393f9711700180ce7e07c43b51@333.333.333.333 (55) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 6: Contact: (46) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 7: Server: MERA MSIP v.3.0 (23) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 8: Content-Type: application/sdp (29) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 9: Content-Length: 241 (21) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Header 10: (0) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: v=0 (3) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: o=- 1175586664 1175586664 IN IP4 222.222.222.222 (47) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: s=- (3) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: c=IN IP4 222.222.222.222 (23) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: t=0 0 (5) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: m=audio 22480 RTP/AVP 18 0 101 (30) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Apr 3 03:51:04] VERBOSE[20549] logger.c: --- (10 headers 11 lines) --- [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6331a7393f9711700180ce7e07c43b51@333.333.333.333' Request 102: Found [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: SIP response 183 to standard invite [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found RTP audio format 18 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found RTP audio format 0 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found RTP audio format 101 [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: Peer doesn't provide T.38 UDPTL [Apr 3 03:51:04] VERBOSE[20549] logger.c: Peer audio RTP is at port 222.222.222.222:22480 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found description format G729 for ID 18 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Got unsupported a:fmtp in SDP offer [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found description format PCMU for ID 0 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Found description format telephone-event for ID 101 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Got unsupported a:fmtp in SDP offer [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: T38 state changed to 0 on channel SIP/provider-09d35848 [Apr 3 03:51:04] VERBOSE[20549] logger.c: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) [Apr 3 03:51:04] VERBOSE[20549] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Apr 3 03:51:04] VERBOSE[20549] logger.c: Peer audio RTP is at port 222.222.222.222:22480 [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: We're settling with these formats: 0x104 (ulaw|g729) [Apr 3 03:51:04] DEBUG[20549] chan_sip.c: We have an owner, now see if we need to change this call [Apr 3 03:51:04] VERBOSE[20557] logger.c: -- Call on SIP/provider-09d35848 left from hold [Apr 3 03:51:04] VERBOSE[20557] logger.c: -- SIP/provider-09d35848 is making progress passing it to SIP/111.111.111.111-09d218f0 [Apr 3 03:51:04] DEBUG[20557] rtp.c: Setting early bridge SDP of 'SIP/111.111.111.111-09d218f0' with that of 'SIP/provider-09d35848'