<--- SIP read from 67.55.159.160:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- <--- SIP read from 192.168.2.5:5064 ---> INVITE sip:17201234567@192.168.2.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bKdc6d822cec6a093c From: "Ext 2201" ;tag=a812cbed43f79660 To: Contact: Supported: replaces, timer, path Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11421 INVITE User-Agent: Grandstream GXP2000 1.1.2.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 347 v=0 o=2201 8000 8000 IN IP4 192.168.2.5 s=SIP Call c=IN IP4 192.168.2.5 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 99 9 3 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/16000 a=rtpmap:3 GSM/8000 a=ptime:20 <-------------> --- (13 headers 17 lines) --- Sending to 192.168.2.5 : 5064 (no NAT) Using INVITE request as basis request - e08be4896afb71bd@192.168.2.5 <--- Reliably Transmitting (no NAT) to 192.168.2.5:5064 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bKdc6d822cec6a093c;received=192.168.2.5 From: "Ext 2201" ;tag=a812cbed43f79660 To: ;tag=as6b0a727b Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11421 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71faa331" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e08be4896afb71bd@192.168.2.5' in 32000 ms (Method: INVITE) Found user '2201' <--- SIP read from 192.168.2.5:5064 ---> ACK sip:17201234567@192.168.2.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bKdc6d822cec6a093c From: "Ext 2201" ;tag=a812cbed43f79660 To: ;tag=as6b0a727b Contact: Supported: path Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11421 ACK User-Agent: Grandstream GXP2000 1.1.2.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from 192.168.2.5:5064 ---> INVITE sip:17201234567@192.168.2.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f From: "Ext 2201" ;tag=a812cbed43f79660 To: Contact: Supported: replaces, timer, path Proxy-Authorization: Digest username="2201", realm="asterisk", algorithm=MD5, uri="sip:17201234567@192.168.2.201", nonce="71faa331", response="59b1507944c33c5b7b45693723ad1354" Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11422 INVITE User-Agent: Grandstream GXP2000 1.1.2.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 347 v=0 o=2201 8000 8001 IN IP4 192.168.2.5 s=SIP Call c=IN IP4 192.168.2.5 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 99 9 3 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=rtpmap:9 G722/16000 a=rtpmap:3 GSM/8000 a=ptime:20 <-------------> --- (14 headers 17 lines) --- Sending to 192.168.2.5 : 5064 (no NAT) Using INVITE request as basis request - e08be4896afb71bd@192.168.2.5 Found user '2201' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 99 Found RTP audio format 9 Found RTP audio format 3 Peer audio RTP is at port 192.168.2.5:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format G726-32 for ID 2 Found description format iLBC for ID 99 Found description format G722 for ID 9 Found description format GSM for ID 3 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.2.5:5004 Looking for 17201234567 in home (domain 192.168.2.201) list_route: hop: <--- Transmitting (no NAT) to 192.168.2.5:5064 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f;received=192.168.2.5 From: "Ext 2201" ;tag=a812cbed43f79660 To: Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11422 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [17201234567@home:1] Dial("SIP/2201-081e77d8", "SIP/lesnet_out/17201234567") in new stack [Apr 13 12:38:50] WARNING[26210]: chan_sip.c:2737 create_addr: No such host: lesnet_out Really destroying SIP dialog '6f9e474747a255d45fa8e15d7f500ecf@192.168.2.201' Method: INVITE [Apr 13 12:38:50] WARNING[26210]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [17201234567@home:2] SayDigits("SIP/2201-081e77d8", "17201234567") in new stack Audio is at 192.168.2.201 port 17244 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP <--- Transmitting (no NAT) to 192.168.2.5:5064 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f;received=192.168.2.5 From: "Ext 2201" ;tag=a812cbed43f79660 To: ;tag=as6cf07fbc Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11422 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 233 v=0 o=root 26180 26180 IN IP4 192.168.2.201 s=session c=IN IP4 192.168.2.201 t=0 0 m=audio 17244 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Playing 'digits/1' (language 'en') -- Playing 'digits/7' (language 'en') Really destroying SIP dialog '6929d341645213f668a2f842313f8116@192.168.2.201' Method: REGISTER Really destroying SIP dialog '1f88d90b32e157b5478dbc2a41e99194@192.168.2.201' Method: REGISTER -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/9' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/8' (language 'en') <--- SIP read from 192.168.2.5:5064 ---> CANCEL sip:17201234567@192.168.2.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f From: "Ext 2201" ;tag=a812cbed43f79660 To: Supported: path Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11422 CANCEL User-Agent: Grandstream GXP2000 1.1.2.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.2.5 : 5064 (no NAT) <--- Reliably Transmitting (no NAT) to 192.168.2.5:5064 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f;received=192.168.2.5 From: "Ext 2201" ;tag=a812cbed43f79660 To: ;tag=as6cf07fbc Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11422 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.2.5:5064 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f;received=192.168.2.5 From: "Ext 2201" ;tag=a812cbed43f79660 To: ;tag=as6cf07fbc Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11422 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (home, 17201234567, 2) exited non-zero on 'SIP/2201-081e77d8' [Apr 13 12:38:55] ERROR[26210]: cdr_addon_mysql.c:148 mysql_log: cdr_mysql: cannot connect to database server localhost. <--- SIP read from 192.168.2.5:5064 ---> ACK sip:17201234567@192.168.2.201 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.5:5064;branch=z9hG4bK3e7b1b5f723d089f From: "Ext 2201" ;tag=a812cbed43f79660 To: ;tag=as6cf07fbc Contact: Supported: path Proxy-Authorization: Digest username="2201", realm="asterisk", algorithm=MD5, uri="sip:17201234567@192.168.2.201", nonce="71faa331", response="59b1507944c33c5b7b45693723ad1354" Call-ID: e08be4896afb71bd@192.168.2.5 CSeq: 11422 ACK User-Agent: Grandstream GXP2000 1.1.2.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog 'e08be4896afb71bd@192.168.2.5' Method: ACK <--- SIP read from 67.55.159.160:5060 ---> <------------->