[Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: <--- SIP read from UDP:XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@YYY.YYY.YYY.3YY SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK65926183;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as5afcb650 To: Contact: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Tue, 05 Oct 2010 18:32:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 304 v=0 o=root 10383783 10383783 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 15492 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 5 21:36:24] VERBOSE[3458] chan_sip.c: --- (14 headers 13 lines) --- [Oct 5 21:36:24] VERBOSE[3458] netsock.c: == Using SIP RTP CoS mark 5 [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Sending to XXX.XXX.XXX.2XX : 5060 (no NAT) [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Using INVITE request as basis request - 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Found peer 'EXT-SRVARG009' for '5521111111' from XXX.XXX.XXX.2XX:5060 [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Found RTP audio format 0 [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Found RTP audio format 8 [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Found RTP audio format 101 [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Found audio description format PCMU for ID 0 [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Peer audio RTP is at port XXX.XXX.XXX.2XX:15492 [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: Looking for 545454 in RPInbound (domain YYY.YYY.YYY.3YY) [Oct 5 21:36:24] VERBOSE[3458] chan_sip.c: list_route: hop: [Oct 5 21:36:24] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 5 21:36:24] VERBOSE[10748] pbx.c: -- Executing [545454@RPInbound:1] Playback("SIP/EXT-SRVARG009-00002775", "we-dont-have-tech-support") in new stack [Oct 5 21:36:24] VERBOSE[10748] chan_sip.c: Audio is at YYY.YYY.YYY.3YY port 15372 [Oct 5 21:36:24] VERBOSE[10748] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 5 21:36:24] VERBOSE[10748] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 5 21:36:24] VERBOSE[10748] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 5 21:36:24] VERBOSE[10748] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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: 302 v=0 o=root 36273902 36273902 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15372 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 5 21:36:24] VERBOSE[10748] file.c: -- Playing 'we-dont-have-tech-support.ulaw' (language 'en') [Oct 5 21:36:25] VERBOSE[3458] chan_sip.c: <--- SIP read from UDP:XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@YYY.YYY.YYY.3YY SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK65926183;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as5afcb650 To: Contact: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Tue, 05 Oct 2010 18:32:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 304 v=0 o=root 10383783 10383783 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 15492 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 5 21:36:25] VERBOSE[3458] chan_sip.c: --- (14 headers 13 lines) --- [Oct 5 21:36:25] VERBOSE[3458] chan_sip.c: Ignoring this INVITE request [Oct 5 21:36:25] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 5 21:36:25] VERBOSE[3458] chan_sip.c: Audio is at YYY.YYY.YYY.3YY port 15372 [Oct 5 21:36:25] VERBOSE[3458] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 5 21:36:25] VERBOSE[3458] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 5 21:36:25] VERBOSE[3458] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 5 21:36:25] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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: 302 v=0 o=root 36273902 36273903 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15372 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 5 21:36:25] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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: 302 v=0 o=root 36273902 36273902 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15372 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 5 21:36:26] VERBOSE[3458] chan_sip.c: <--- SIP read from UDP:XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@YYY.YYY.YYY.3YY SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK65926183;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as5afcb650 To: Contact: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Tue, 05 Oct 2010 18:32:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 304 v=0 o=root 10383783 10383783 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 15492 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 5 21:36:26] VERBOSE[3458] chan_sip.c: --- (14 headers 13 lines) --- [Oct 5 21:36:26] VERBOSE[3458] chan_sip.c: Ignoring this INVITE request [Oct 5 21:36:26] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 5 21:36:26] VERBOSE[3458] chan_sip.c: Audio is at YYY.YYY.YYY.3YY port 15372 [Oct 5 21:36:26] VERBOSE[3458] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 5 21:36:26] VERBOSE[3458] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 5 21:36:26] VERBOSE[3458] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 5 21:36:26] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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: 302 v=0 o=root 36273902 36273904 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15372 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 5 21:36:26] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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: 302 v=0 o=root 36273902 36273902 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15372 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 5 21:36:26] VERBOSE[10748] pbx.c: -- Auto fallthrough, channel 'SIP/EXT-SRVARG009-00002775' status is 'UNKNOWN' [Oct 5 21:36:26] VERBOSE[10748] chan_sip.c: Scheduling destruction of SIP dialog '48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX' in 32000 ms (Method: INVITE) [Oct 5 21:36:28] VERBOSE[3458] chan_sip.c: <--- SIP read from UDP:XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@YYY.YYY.YYY.3YY SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK65926183;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as5afcb650 To: Contact: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Tue, 05 Oct 2010 18:32:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 304 v=0 o=root 10383783 10383783 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 15492 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 5 21:36:28] VERBOSE[3458] chan_sip.c: --- (14 headers 13 lines) --- [Oct 5 21:36:28] VERBOSE[3458] chan_sip.c: Ignoring this INVITE request [Oct 5 21:36:28] NOTICE[3458] chan_sip.c: Unable to create/find SIP channel for this INVITE [Oct 5 21:36:28] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Content-Length: 0 <------------> [Oct 5 21:36:28] VERBOSE[3458] chan_sip.c: Scheduling destruction of SIP dialog '48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX' in 32000 ms (Method: INVITE) [Oct 5 21:36:28] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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: 302 v=0 o=root 36273902 36273902 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15372 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 5 21:36:32] VERBOSE[3458] chan_sip.c: <--- SIP read from UDP:XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@YYY.YYY.YYY.3YY SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK65926183;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as5afcb650 To: Contact: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Tue, 05 Oct 2010 18:32:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 304 v=0 o=root 10383783 10383783 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 15492 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 5 21:36:32] VERBOSE[3458] chan_sip.c: --- (14 headers 13 lines) --- [Oct 5 21:36:32] VERBOSE[3458] chan_sip.c: Ignoring this INVITE request [Oct 5 21:36:32] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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: 302 v=0 o=root 36273902 36273902 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15372 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 5 21:36:36] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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: 302 v=0 o=root 36273902 36273902 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15372 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 5 21:36:40] VERBOSE[3458] chan_sip.c: <--- SIP read from UDP:XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@YYY.YYY.YYY.3YY SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK65926183;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as5afcb650 To: Contact: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Tue, 05 Oct 2010 18:32:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 304 v=0 o=root 10383783 10383783 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 15492 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 5 21:36:40] VERBOSE[3458] chan_sip.c: --- (14 headers 13 lines) --- [Oct 5 21:36:40] VERBOSE[3458] chan_sip.c: Ignoring this INVITE request [Oct 5 21:36:40] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as2e4b9a75 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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: 302 v=0 o=root 36273902 36273902 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15372 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 5 21:36:44] WARNING[3458] chan_sip.c: Maximum retries exceeded on transmission 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: <--- SIP read from UDP:XXX.XXX.XXX.2XX:5060 ---> INVITE sip:545454@YYY.YYY.YYY.3YY SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.2XX:5060;branch=z9hG4bK65926183;rport Max-Forwards: 70 From: "Bere, Marcelo" ;tag=as5afcb650 To: Contact: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE User-Agent: MyCompany Gateway Date: Tue, 05 Oct 2010 18:32:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 304 v=0 o=root 10383783 10383783 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 15492 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 5 21:36:56] VERBOSE[3458] chan_sip.c: --- (14 headers 13 lines) --- [Oct 5 21:36:56] VERBOSE[3458] netsock.c: == Using SIP RTP CoS mark 5 [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Sending to XXX.XXX.XXX.2XX : 5060 (no NAT) [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Using INVITE request as basis request - 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Found peer 'EXT-SRVARG009' for '5521111111' from XXX.XXX.XXX.2XX:5060 [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Found RTP audio format 0 [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Found RTP audio format 8 [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Found RTP audio format 101 [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Found audio description format PCMU for ID 0 [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Found audio description format PCMA for ID 8 [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Found audio description format telephone-event for ID 101 [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Peer audio RTP is at port XXX.XXX.XXX.2XX:15492 [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: Looking for 545454 in RPInbound (domain YYY.YYY.YYY.3YY) [Oct 5 21:36:56] VERBOSE[3458] chan_sip.c: list_route: hop: [Oct 5 21:36:56] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 5 21:36:56] VERBOSE[10776] pbx.c: -- Executing [545454@RPInbound:1] Playback("SIP/EXT-SRVARG009-00002776", "we-dont-have-tech-support") in new stack [Oct 5 21:36:56] VERBOSE[10776] chan_sip.c: Audio is at YYY.YYY.YYY.3YY port 15670 [Oct 5 21:36:56] VERBOSE[10776] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Oct 5 21:36:56] VERBOSE[10776] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Oct 5 21:36:56] VERBOSE[10776] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Oct 5 21:36:56] VERBOSE[10776] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as6f1fb739 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 719511998 719511998 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15670 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 5 21:36:56] VERBOSE[10776] file.c: -- Playing 'we-dont-have-tech-support.ulaw' (language 'en') [Oct 5 21:36:57] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as6f1fb739 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 719511998 719511998 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15670 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 5 21:36:58] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as6f1fb739 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 719511998 719511998 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15670 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 5 21:36:58] VERBOSE[10776] pbx.c: -- Auto fallthrough, channel 'SIP/EXT-SRVARG009-00002776' status is 'UNKNOWN' [Oct 5 21:36:58] VERBOSE[10776] chan_sip.c: Scheduling destruction of SIP dialog '48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX' in 32000 ms (Method: INVITE) [Oct 5 21:37:00] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as6f1fb739 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 719511998 719511998 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15670 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 5 21:37:04] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as6f1fb739 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 719511998 719511998 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15670 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 5 21:37:08] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as6f1fb739 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 719511998 719511998 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15670 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 5 21:37:12] VERBOSE[3458] chan_sip.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=z9hG4bK65926183;received=XXX.XXX.XXX.2XX;rport=5060 From: "Bere, Marcelo" ;tag=as5afcb650 To: ;tag=as6f1fb739 Call-ID: 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX CSeq: 102 INVITE Server: TeleCompanyIVRv1 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 719511998 719511998 IN IP4 YYY.YYY.YYY.3YY s=Asterisk PBX SVN-branch-1.6.2-r281912 c=IN IP4 YYY.YYY.YYY.3YY t=0 0 m=audio 15670 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 5 21:37:16] WARNING[3458] chan_sip.c: Maximum retries exceeded on transmission 48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 5 21:37:16] VERBOSE[3458] chan_sip.c: Really destroying SIP dialog '48b00c42375588c6146f867f46fb30b7@XXX.XXX.XXX.2XX' Method: INVITE