[Oct 1 20:30:52] VERBOSE[4293] logger.c: <--- SIP read from UDP://XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@ZZZ.ZZZ.ZZZ.3ZZ SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as44a708c7 To: Contact: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Fri, 01 Oct 2010 17:27:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 429761976 429761976 IN IP4 XXX.XXX.XXX.2XX s=Asterisk PBX SVN-branch-1.6.2-r272459 c=IN IP4 XXX.XXX.XXX.2XX t=0 0 m=audio 18942 RTP/AVP 0 8 101 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 <-------------> [Oct 1 20:30:52] VERBOSE[4293] logger.c: --- (14 headers 13 lines) --- [Oct 1 20:30:52] VERBOSE[4293] logger.c: == Using SIP RTP CoS mark 5 [Oct 1 20:30:52] VERBOSE[4293] logger.c: Sending to XXX.XXX.XXX.2XX : 5060 (no NAT) [Oct 1 20:30:52] VERBOSE[4293] logger.c: Using INVITE request as basis request - 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX [Oct 1 20:30:52] VERBOSE[4293] logger.c: No user '21111111' in SIP users list [Oct 1 20:30:52] VERBOSE[4293] logger.c: Found peer 'EXT-SRVARG009' for '21111111' from XXX.XXX.XXX.2XX:5060 [Oct 1 20:30:52] VERBOSE[4293] logger.c: Found RTP audio format 0 [Oct 1 20:30:52] VERBOSE[4293] logger.c: Found RTP audio format 8 [Oct 1 20:30:52] VERBOSE[4293] logger.c: Found RTP audio format 101 [Oct 1 20:30:52] VERBOSE[4293] logger.c: Found audio description format PCMU for ID 0 [Oct 1 20:30:52] VERBOSE[4293] logger.c: Found audio description format PCMA for ID 8 [Oct 1 20:30:52] VERBOSE[4293] logger.c: Found audio description format telephone-event for ID 101 [Oct 1 20:30:52] VERBOSE[4293] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 1 20:30:52] VERBOSE[4293] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Oct 1 20:30:52] VERBOSE[4293] logger.c: Peer audio RTP is at port XXX.XXX.XXX.2XX:18942 [Oct 1 20:30:52] VERBOSE[4293] logger.c: Looking for 545454 in RPInbound (domain ZZZ.ZZZ.ZZZ.3ZZ) [Oct 1 20:30:52] VERBOSE[4293] logger.c: list_route: hop: [Oct 1 20:30:52] VERBOSE[4293] logger.c: <--- Transmitting (no NAT) to XXX.XXX.XXX.2XX:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Oct 1 20:30:52] VERBOSE[12042] logger.c: -- Executing [545454@RPInbound:1] Playback("SIP/EXT-SRVARG009-0004a830", "we-dont-have-tech-support") in new stack [Oct 1 20:30:52] VERBOSE[12042] logger.c: Audio is at ZZZ.ZZZ.ZZZ.3ZZ port 19444 [Oct 1 20:30:52] VERBOSE[12042] logger.c: Adding codec 0x4 (ulaw) to SDP [Oct 1 20:30:52] VERBOSE[12042] logger.c: Adding codec 0x8 (alaw) to SDP [Oct 1 20:30:52] VERBOSE[12042] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 1 20:30:52] VERBOSE[12042] logger.c: <--- Reliably Transmitting (no NAT) to XXX.XXX.XXX.2XX:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 81649272 81649272 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 19444 RTP/AVP 0 8 101 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 <------------> [Oct 1 20:30:53] VERBOSE[12042] logger.c: -- Playing 'we-dont-have-tech-support.slin' (language 'en') [Oct 1 20:30:53] VERBOSE[4293] logger.c: <--- SIP read from UDP://XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@ZZZ.ZZZ.ZZZ.3ZZ SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as44a708c7 To: Contact: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Fri, 01 Oct 2010 17:27:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 429761976 429761976 IN IP4 XXX.XXX.XXX.2XX s=Asterisk PBX SVN-branch-1.6.2-r272459 c=IN IP4 XXX.XXX.XXX.2XX t=0 0 m=audio 18942 RTP/AVP 0 8 101 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 <-------------> [Oct 1 20:30:53] VERBOSE[4293] logger.c: --- (14 headers 13 lines) --- [Oct 1 20:30:53] VERBOSE[4293] logger.c: Ignoring this INVITE request [Oct 1 20:30:53] VERBOSE[4293] logger.c: <--- Transmitting (no NAT) to XXX.XXX.XXX.2XX:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Oct 1 20:30:53] VERBOSE[4293] logger.c: Audio is at ZZZ.ZZZ.ZZZ.3ZZ port 19444 [Oct 1 20:30:53] VERBOSE[4293] logger.c: Adding codec 0x4 (ulaw) to SDP [Oct 1 20:30:53] VERBOSE[4293] logger.c: Adding codec 0x8 (alaw) to SDP [Oct 1 20:30:53] VERBOSE[4293] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 1 20:30:53] VERBOSE[4293] logger.c: <--- Transmitting (no NAT) to XXX.XXX.XXX.2XX:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 81649272 81649273 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 19444 RTP/AVP 0 8 101 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 <------------> [Oct 1 20:30:53] VERBOSE[4293] logger.c: Retransmitting #1 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 81649272 81649272 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 19444 RTP/AVP 0 8 101 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 --- [Oct 1 20:30:54] VERBOSE[4293] logger.c: <--- SIP read from UDP://XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@ZZZ.ZZZ.ZZZ.3ZZ SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as44a708c7 To: Contact: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Fri, 01 Oct 2010 17:27:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 429761976 429761976 IN IP4 XXX.XXX.XXX.2XX s=Asterisk PBX SVN-branch-1.6.2-r272459 c=IN IP4 XXX.XXX.XXX.2XX t=0 0 m=audio 18942 RTP/AVP 0 8 101 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 <-------------> [Oct 1 20:30:54] VERBOSE[4293] logger.c: --- (14 headers 13 lines) --- [Oct 1 20:30:54] VERBOSE[4293] logger.c: Ignoring this INVITE request [Oct 1 20:30:54] VERBOSE[4293] logger.c: <--- Transmitting (no NAT) to XXX.XXX.XXX.2XX:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Oct 1 20:30:54] VERBOSE[4293] logger.c: Audio is at ZZZ.ZZZ.ZZZ.3ZZ port 19444 [Oct 1 20:30:54] VERBOSE[4293] logger.c: Adding codec 0x4 (ulaw) to SDP [Oct 1 20:30:54] VERBOSE[4293] logger.c: Adding codec 0x8 (alaw) to SDP [Oct 1 20:30:54] VERBOSE[4293] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 1 20:30:54] VERBOSE[4293] logger.c: <--- Transmitting (no NAT) to XXX.XXX.XXX.2XX:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 81649272 81649274 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 19444 RTP/AVP 0 8 101 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 <------------> [Oct 1 20:30:54] VERBOSE[4293] logger.c: Retransmitting #2 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 81649272 81649272 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 19444 RTP/AVP 0 8 101 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 --- [Oct 1 20:30:55] VERBOSE[12042] logger.c: -- Auto fallthrough, channel 'SIP/EXT-SRVARG009-0004a830' status is 'UNKNOWN' [Oct 1 20:30:55] VERBOSE[12042] logger.c: Scheduling destruction of SIP dialog '173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX' in 32000 ms (Method: INVITE) [Oct 1 20:30:56] VERBOSE[4293] logger.c: <--- SIP read from UDP://XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@ZZZ.ZZZ.ZZZ.3ZZ SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as44a708c7 To: Contact: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Fri, 01 Oct 2010 17:27:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 429761976 429761976 IN IP4 XXX.XXX.XXX.2XX s=Asterisk PBX SVN-branch-1.6.2-r272459 c=IN IP4 XXX.XXX.XXX.2XX t=0 0 m=audio 18942 RTP/AVP 0 8 101 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 <-------------> [Oct 1 20:30:56] VERBOSE[4293] logger.c: --- (14 headers 13 lines) --- [Oct 1 20:30:56] VERBOSE[4293] logger.c: Ignoring this INVITE request [Oct 1 20:30:56] NOTICE[4293] chan_sip.c: Unable to create/find SIP channel for this INVITE [Oct 1 20:30:56] VERBOSE[4293] logger.c: <--- Transmitting (no NAT) to XXX.XXX.XXX.2XX:5060 ---> SIP/2.0 503 Unavailable Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Content-Length: 0 <------------> [Oct 1 20:30:56] VERBOSE[4293] logger.c: Scheduling destruction of SIP dialog '173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX' in 32000 ms (Method: INVITE) [Oct 1 20:30:56] VERBOSE[4293] logger.c: Retransmitting #3 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 81649272 81649272 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 19444 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:00] VERBOSE[4293] logger.c: <--- SIP read from UDP://XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@ZZZ.ZZZ.ZZZ.3ZZ SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as44a708c7 To: Contact: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Fri, 01 Oct 2010 17:27:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 429761976 429761976 IN IP4 XXX.XXX.XXX.2XX s=Asterisk PBX SVN-branch-1.6.2-r272459 c=IN IP4 XXX.XXX.XXX.2XX t=0 0 m=audio 18942 RTP/AVP 0 8 101 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 <-------------> [Oct 1 20:31:00] VERBOSE[4293] logger.c: --- (14 headers 13 lines) --- [Oct 1 20:31:00] VERBOSE[4293] logger.c: Ignoring this INVITE request [Oct 1 20:31:00] VERBOSE[4293] logger.c: Retransmitting #4 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 81649272 81649272 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 19444 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:04] VERBOSE[4293] logger.c: Retransmitting #5 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 81649272 81649272 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 19444 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:08] VERBOSE[4293] logger.c: Retransmitting #6 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as12241245 Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 304 v=0 o=root 81649272 81649272 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 19444 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:08] VERBOSE[4293] logger.c: <--- SIP read from UDP://XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@ZZZ.ZZZ.ZZZ.3ZZ SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as44a708c7 To: Contact: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Fri, 01 Oct 2010 17:27:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 429761976 429761976 IN IP4 XXX.XXX.XXX.2XX s=Asterisk PBX SVN-branch-1.6.2-r272459 c=IN IP4 XXX.XXX.XXX.2XX t=0 0 m=audio 18942 RTP/AVP 0 8 101 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 <-------------> [Oct 1 20:31:08] VERBOSE[4293] logger.c: --- (14 headers 13 lines) --- [Oct 1 20:31:08] VERBOSE[4293] logger.c: Ignoring this INVITE request [Oct 1 20:31:12] WARNING[4293] chan_sip.c: Maximum retries exceeded on transmission 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 1 20:31:24] VERBOSE[4293] logger.c: <--- SIP read from UDP://XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@ZZZ.ZZZ.ZZZ.3ZZ SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as44a708c7 To: Contact: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Fri, 01 Oct 2010 17:27:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 306 v=0 o=root 429761976 429761976 IN IP4 XXX.XXX.XXX.2XX s=Asterisk PBX SVN-branch-1.6.2-r272459 c=IN IP4 XXX.XXX.XXX.2XX t=0 0 m=audio 18942 RTP/AVP 0 8 101 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 <-------------> [Oct 1 20:31:24] VERBOSE[4293] logger.c: --- (14 headers 13 lines) --- [Oct 1 20:31:24] VERBOSE[4293] logger.c: == Using SIP RTP CoS mark 5 [Oct 1 20:31:24] VERBOSE[4293] logger.c: Sending to XXX.XXX.XXX.2XX : 5060 (no NAT) [Oct 1 20:31:24] VERBOSE[4293] logger.c: Using INVITE request as basis request - 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX [Oct 1 20:31:24] VERBOSE[4293] logger.c: No user '21111111' in SIP users list [Oct 1 20:31:24] VERBOSE[4293] logger.c: Found peer 'EXT-SRVARG009' for '21111111' from XXX.XXX.XXX.2XX:5060 [Oct 1 20:31:24] VERBOSE[4293] logger.c: Found RTP audio format 0 [Oct 1 20:31:24] VERBOSE[4293] logger.c: Found RTP audio format 8 [Oct 1 20:31:24] VERBOSE[4293] logger.c: Found RTP audio format 101 [Oct 1 20:31:24] VERBOSE[4293] logger.c: Found audio description format PCMU for ID 0 [Oct 1 20:31:24] VERBOSE[4293] logger.c: Found audio description format PCMA for ID 8 [Oct 1 20:31:24] VERBOSE[4293] logger.c: Found audio description format telephone-event for ID 101 [Oct 1 20:31:24] VERBOSE[4293] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 1 20:31:24] VERBOSE[4293] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Oct 1 20:31:24] VERBOSE[4293] logger.c: Peer audio RTP is at port XXX.XXX.XXX.2XX:18942 [Oct 1 20:31:24] VERBOSE[4293] logger.c: Looking for 545454 in RPInbound (domain ZZZ.ZZZ.ZZZ.3ZZ) [Oct 1 20:31:24] VERBOSE[4293] logger.c: list_route: hop: [Oct 1 20:31:24] VERBOSE[4293] logger.c: <--- Transmitting (no NAT) to XXX.XXX.XXX.2XX:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> [Oct 1 20:31:24] VERBOSE[12069] logger.c: -- Executing [545454@RPInbound:1] Playback("SIP/EXT-SRVARG009-0004a831", "we-dont-have-tech-support") in new stack [Oct 1 20:31:24] VERBOSE[12069] logger.c: Audio is at ZZZ.ZZZ.ZZZ.3ZZ port 10090 [Oct 1 20:31:24] VERBOSE[12069] logger.c: Adding codec 0x4 (ulaw) to SDP [Oct 1 20:31:24] VERBOSE[12069] logger.c: Adding codec 0x8 (alaw) to SDP [Oct 1 20:31:24] VERBOSE[12069] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 1 20:31:24] VERBOSE[12069] logger.c: <--- Reliably Transmitting (no NAT) to XXX.XXX.XXX.2XX:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as57e9873d Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 308 v=0 o=root 2085537887 2085537887 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 10090 RTP/AVP 0 8 101 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 <------------> [Oct 1 20:31:25] VERBOSE[12069] logger.c: -- Playing 'we-dont-have-tech-support.slin' (language 'en') [Oct 1 20:31:25] VERBOSE[4293] logger.c: Retransmitting #1 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as57e9873d Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 308 v=0 o=root 2085537887 2085537887 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 10090 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:26] VERBOSE[4293] logger.c: Retransmitting #2 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as57e9873d Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 308 v=0 o=root 2085537887 2085537887 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 10090 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:27] VERBOSE[12069] logger.c: -- Auto fallthrough, channel 'SIP/EXT-SRVARG009-0004a831' status is 'UNKNOWN' [Oct 1 20:31:27] VERBOSE[12069] logger.c: Scheduling destruction of SIP dialog '173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX' in 32000 ms (Method: INVITE) [Oct 1 20:31:28] WARNING[4293] chan_sip.c: Trying to destroy "173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX", not found in dialog list?!?! [Oct 1 20:31:28] VERBOSE[4293] logger.c: Retransmitting #3 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as57e9873d Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 308 v=0 o=root 2085537887 2085537887 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 10090 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:32] VERBOSE[4293] logger.c: Retransmitting #4 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as57e9873d Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 308 v=0 o=root 2085537887 2085537887 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 10090 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:36] VERBOSE[4293] logger.c: Retransmitting #5 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as57e9873d Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 308 v=0 o=root 2085537887 2085537887 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 10090 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:40] VERBOSE[4293] logger.c: Retransmitting #6 (no NAT) to XXX.XXX.XXX.2XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK02d4fb29;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as44a708c7 To: ;tag=as57e9873d Call-ID: 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: CompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 308 v=0 o=root 2085537887 2085537887 IN IP4 ZZZ.ZZZ.ZZZ.3ZZ s=Asterisk PBX SVN-branch-1.6.0-r250265 c=IN IP4 ZZZ.ZZZ.ZZZ.3ZZ t=0 0 m=audio 10090 RTP/AVP 0 8 101 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 --- [Oct 1 20:31:44] WARNING[4293] chan_sip.c: Maximum retries exceeded on transmission 173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 1 20:31:44] VERBOSE[4293] logger.c: Really destroying SIP dialog '173bbb9e6a1228ae2f649d1622840608@XXX.XXX.XXX.2XX' Method: INVITE