<--- SIP read from 10.3.1.31:5060 ---> INVITE sip:500@10.3.1.31:5061;transport=udp SIP/2.0 Record-Route: Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bKf495.30eae2c7.0 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK150FBDA86B8E48E89CC7577574F7C2C2 From: Patrick Baker ;tag=1605889415 To: Contact: Call-ID: 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 CSeq: 9634 INVITE Max-Forwards: 69 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 253 v=0 o=pbaker2 3065208534 3065208704 IN IP4 10.3.1.115 s=X-Lite c=IN IP4 10.3.1.31 t=0 0 m=audio 35188 RTP/AVP 0 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=nortpproxy:yes <-------------> --- (13 headers 12 lines) --- == Using TOS bits 0 == Using CoS mark 5 Sending to 10.3.1.31 : 5060 (NAT) Using INVITE request as basis request - 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 No user 'pbaker2' in SIP users list Found peer 'openser' for 'pbaker2' from 10.3.1.31:5060 Found RTP audio format 0 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 10.3.1.31:35188 Found audio description format pcmu for ID 0 Found audio description format speex for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140), peer - audio=0x204 (ulaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x204 (ulaw|speex) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.3.1.31:35188 Looking for 500 in default (domain 10.3.1.31) list_route: hop: <--- Transmitting (no NAT) to 10.3.1.31:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bKf495.30eae2c7.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK150FBDA86B8E48E89CC7577574F7C2C2 Record-Route: From: Patrick Baker ;tag=1605889415 To: Call-ID: 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 CSeq: 9634 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [500@default:1] Playback("SIP/10.3.1.31-081c4a28", "demo-abouttotry") in new stack Audio is at xxx.206.xxx.136 port 52280 Adding codec 0x4 (ulaw) to SDP Adding codec 0x200 (speex) to SDP Adding non-codec 0x1 (telephone-event) to SDP phonesys-slave*CLI> <--- Reliably Transmitting (no NAT) to 10.3.1.31:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bKf495.30eae2c7.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK150FBDA86B8E48E89CC7577574F7C2C2 Record-Route: From: Patrick Baker ;tag=1605889415 To: ;tag=as674bae5c Call-ID: 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 CSeq: 9634 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 1112739693 1112739693 IN IP4 xxx.206.xxx.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 xxx.206.xxx.136 t=0 0 m=audio 52280 RTP/AVP 0 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Playing 'demo-abouttotry.slin' (language 'en') Retransmitting #1 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bKf495.30eae2c7.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK150FBDA86B8E48E89CC7577574F7C2C2 Record-Route: From: Patrick Baker ;tag=1605889415 To: ;tag=as674bae5c Call-ID: 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 CSeq: 9634 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 1112739693 1112739693 IN IP4 xxx.206.xxx.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 xxx.206.xxx.136 t=0 0 m=audio 52280 RTP/AVP 0 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bKf495.30eae2c7.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK150FBDA86B8E48E89CC7577574F7C2C2 Record-Route: From: Patrick Baker ;tag=1605889415 To: ;tag=as674bae5c Call-ID: 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 CSeq: 9634 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 1112739693 1112739693 IN IP4 xxx.206.xxx.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 xxx.206.xxx.136 t=0 0 m=audio 52280 RTP/AVP 0 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bKf495.30eae2c7.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK150FBDA86B8E48E89CC7577574F7C2C2 Record-Route: From: Patrick Baker ;tag=1605889415 To: ;tag=as674bae5c Call-ID: 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 CSeq: 9634 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 1112739693 1112739693 IN IP4 xxx.206.xxx.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 xxx.206.xxx.136 t=0 0 m=audio 52280 RTP/AVP 0 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #4 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bKf495.30eae2c7.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK150FBDA86B8E48E89CC7577574F7C2C2 Record-Route: From: Patrick Baker ;tag=1605889415 To: ;tag=as674bae5c Call-ID: 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 CSeq: 9634 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 1112739693 1112739693 IN IP4 xxx.206.xxx.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 xxx.206.xxx.136 t=0 0 m=audio 52280 RTP/AVP 0 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #5 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bKf495.30eae2c7.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK150FBDA86B8E48E89CC7577574F7C2C2 Record-Route: From: Patrick Baker ;tag=1605889415 To: ;tag=as674bae5c Call-ID: 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 CSeq: 9634 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 1112739693 1112739693 IN IP4 xxx.206.xxx.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 xxx.206.xxx.136 t=0 0 m=audio 52280 RTP/AVP 0 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Executing [500@default:2] Dial("SIP/10.3.1.31-081c4a28", "IAX2/guest@misery.digium.com/s@default") in new stack -- Called guest@misery.digium.com/s@default -- Call accepted by 216.207.245.8 (format gsm) -- Format for call is gsm -- IAX2/216.207.245.8:4569-1 is ringing [Dec 7 22:22:10] NOTICE[6495]: rtp.c:998 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.3.1.31 -- IAX2/216.207.245.8:4569-1 answered SIP/10.3.1.31-081c4a28 Retransmitting #6 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bKf495.30eae2c7.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK150FBDA86B8E48E89CC7577574F7C2C2 Record-Route: From: Patrick Baker ;tag=1605889415 To: ;tag=as674bae5c Call-ID: 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 CSeq: 9634 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 305 v=0 o=root 1112739693 1112739693 IN IP4 xxx.206.xxx.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 xxx.206.xxx.136 t=0 0 m=audio 52280 RTP/AVP 0 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 7 22:22:15] WARNING[6495]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded on transmission 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 for seqno 9634 (Critical Response) [Dec 7 22:22:15] WARNING[6495]: chan_sip.c:2361 retrans_pkt: Hanging up call 1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115 - no reply to our critical packet. -- Hungup 'IAX2/216.207.245.8:4569-1' == Spawn extension (default, 500, 2) exited non-zero on 'SIP/10.3.1.31-081c4a28' Really destroying SIP dialog '1F66EFCE-ED0E-DE7E-20B0-04F00C39B891@10.3.1.115' Method: INVITE