pbx*CLI> sip debug peer eden-1000a SIP Debugging Enabled for IP: 10.253.4.50:5060 pbx*CLI> <-- SIP read from 10.253.4.50:5060: INVITE sip:9990@pbx.sdnglobal.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF From: "eden-1000a" ;tag=D4964260-95FB99E3 To: CSeq: 1 INVITE Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Using INVITE request as basis request - a857d7ac-36f29d46-4d6ef889@10.253.4.50 Sending to 10.253.4.50 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.253.4.50:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as7f808f0f Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2584558d" Content-Length: 0 --- Scheduling destruction of call 'a857d7ac-36f29d46-4d6ef889@10.253.4.50' in 15000 ms Found user 'eden-1000a' pbx*CLI> <-- SIP read from 10.253.4.50:5060: INVITE sip:9990@pbx.sdnglobal.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF From: "eden-1000a" ;tag=D4964260-95FB99E3 To: CSeq: 1 INVITE Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Ignoring this INVITE request pbx*CLI> <-- SIP read from 10.253.4.50:5060: ACK sip:9990@pbx.sdnglobal.com SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as7f808f0f CSeq: 1 ACK Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- pbx*CLI> <-- SIP read from 10.253.4.50:5060: INVITE sip:9990@pbx.sdnglobal.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA From: "eden-1000a" ;tag=D4964260-95FB99E3 To: CSeq: 2 INVITE Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="eden-1000a", realm="asterisk", nonce="2584558d", uri="sip:9990@pbx.sdnglobal.com;user=phone", response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines) --- Using INVITE request as basis request - a857d7ac-36f29d46-4d6ef889@10.253.4.50 Sending to 10.253.4.50 : 5060 (non-NAT) Found user 'eden-1000a' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.253.4.50:2228 Peer video RTP is at port 10.253.4.50:65535 Found description format PCMU Found description format G729 Found description format PCMA Found description format telephone-event Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 9990 in eden-dialout (domain pbx.sdnglobal.com;user=phone) list_route: hop: Transmitting (no NAT) to 10.253.4.50:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Answer("SIP/eden-1000a-4150cc98", "") in new stack We're at 172.30.42.5 port 29816 Video is at 172.30.42.5 port 29214 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp ontent-Length: 235 v=0 o=root 5641 5641 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing VoiceMailMain("SIP/eden-1000a-4150cc98", "1000@eden") in new stack -- Playing 'vm-password' (language 'en') pbx*CLI> <-- SIP read from 10.253.4.50:5060: INVITE sip:9990@pbx.sdnglobal.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA From: "eden-1000a" ;tag=D4964260-95FB99E3 To: CSeq: 2 INVITE Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="eden-1000a", realm="asterisk", nonce="2584558d", uri="sip:9990@pbx.sdnglobal.com;user=phone", response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines) --- Ignoring this INVITE request We're at 172.30.42.5 port 29816 Video is at 172.30.42.5 port 29214 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp ontent-Length: 235 v=0 o=root 5641 5642 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 10.253.4.50:5060: INVITE sip:9990@pbx.sdnglobal.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA From: "eden-1000a" ;tag=D4964260-95FB99E3 To: CSeq: 2 INVITE Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="eden-1000a", realm="asterisk", nonce="2584558d", uri="sip:9990@pbx.sdnglobal.com;user=phone", response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines) --- Ignoring this INVITE request We're at 172.30.42.5 port 29816 Video is at 172.30.42.5 port 29214 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp ontent-Length: 235 v=0 o=root 5641 5643 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 10.253.4.50:5060: ACK sip:9990@172.30.42.5 SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK1674aeae5EA3A4B From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 CSeq: 2 ACK Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Proxy-Authorization: Digest username="eden-1000a", realm="asterisk", nonce="2584558d", uri="sip:9990@pbx.sdnglobal.com;user=phone", response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines) --- Retransmitting #1 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5641 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5642 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5641 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5642 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/1' (language 'en') Retransmitting #3 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5641 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5642 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'vm-Old' (language 'en') -- Playing 'vm-message' (language 'en') -- Playing 'vm-onefor' (language 'en') -- Playing 'digits/7' (language 'en') -- Playing 'vm-Old' (language 'en') -- Playing 'vm-first' (language 'en') Retransmitting #4 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5641 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'vm-message' (language 'en') Retransmitting #4 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5642 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- == Parsing '/var/spool/asterisk/voicemail/eden/1000/Old/msg0000.txt': Found -- Playing 'vm-received' (language 'en') -- Playing 'digits/at' (language 'en') -- Playing 'digits/17' (language 'en') -- Playing 'digits/hundred' (language 'en') Retransmitting #5 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5641 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- SIP/cp-0821a7d8 is making progress passing it to IAX2/acppbx-102 Retransmitting #5 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5642 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'digits/50' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'hours' (language 'en') -- Playing '/var/spool/asterisk/voicemail/eden/1000/Old/msg0000' (language 'en') 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.253.4.50:5060: OPTIONS sip:eden-1000a@10.253.4.50 SIP/2.0 Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport From: "asterisk" ;tag=as021e29c4 To: Contact: Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@172.30.42.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 20 Mar 2007 23:01:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Retransmitting #6 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5641 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #6 (no NAT) to 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 5641 5642 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 10.253.4.50:5060: OPTIONS sip:eden-1000a@10.253.4.50 SIP/2.0 Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport From: "asterisk" ;tag=as021e29c4 To: Contact: Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@172.30.42.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 20 Mar 2007 23:01:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ontent-Length: 0 --- pbx*CLI> exit <-- SIP read from 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport From: "asterisk" ;tag=as021e29c4 To: ;tag=9E3B7462-6F180925 CSeq: 102 OPTIONS Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@172.30.42.5 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Content-Length: 0 --- (10 headers 0 lines) --- Destroying call '2a1ab9c42b63a0305f6de14715f4f8f4@172.30.42.5' pbx*CLI> exit <-- SIP read from 10.253.4.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport From: "asterisk" ;tag=as021e29c4 To: ;tag=9E3B7462-6F180925 CSeq: 102 OPTIONS Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@172.30.42.5 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Content-Length: 0 --- (10 headers 0 lines) --- Mar 20 18:01:44 WARNING[2770]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission a857d7ac-36f29d46-4d6ef889@10.253.4.50 for seqno 2 (Critical Response) Mar 20 18:01:44 WARNING[2770]: chan_sip.c:1245 retrans_pkt: Hanging up call a857d7ac-36f29d46-4d6ef889@10.253.4.50 - no reply to our critical packet. == Spawn extension (eden-dialout, 9990, 2) exited non-zero on 'SIP/eden-1000a-4150cc98' Mar 20 18:01:45 WARNING[2770]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission a857d7ac-36f29d46-4d6ef889@10.253.4.50 for seqno 2 (Non-critical Response) -- SIP/cp-0821a7d8 answered IAX2/acppbx-102 Destroying call 'a857d7ac-36f29d46-4d6ef889@10.253.4.50' pbx*CLI> exit <-- SIP read from 10.253.4.50:5060: BYE sip:9990@172.30.42.5 SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 CSeq: 3 BYE Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Proxy-Authorization: Digest username="eden-1000a", realm="asterisk", nonce="2584558d", uri="sip:9990@pbx.sdnglobal.com;user=phone", response="32687f30de53796b3ad2c3283d199984", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (NAT) to 10.253.4.50:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- pbx*CLI> exit <-- SIP read from 10.253.4.50:5060: BYE sip:9990@172.30.42.5 SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 CSeq: 3 BYE Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Proxy-Authorization: Digest username="eden-1000a", realm="asterisk", nonce="2584558d", uri="sip:9990@pbx.sdnglobal.com;user=phone", response="32687f30de53796b3ad2c3283d199984", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (NAT) to 10.253.4.50:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C;received=10.253.4.50 From: "eden-1000a" ;tag=D4964260-95FB99E3 To: ;tag=as789e1ad9 Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- pbx*CLI> exit