INVITE sip:7303@10.200.104.112:5061 SIP/2.0 Via: SIP/2.0/TLS 10.200.158.208:5062;branch=z9hG4bK846448099 From: "7309@FLATESTE" ;tag=1080438895 To: Call-ID: 433391560@10.200.158.208 CSeq: 2 INVITE Contact: Authorization: Digest username="7309", realm="asterisk", nonce="585ce05e", uri="sip:7303@10.200.104.112:5061", response="5a54c5269bde165a966cfeddba53548c", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T28P 2.60.0.80 Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 476 v=0 o=- 20002 20002 IN IP4 10.200.158.208 s=SDP data c=IN IP4 10.200.158.208 t=0 0 m=audio 11784 RTP/SAVP 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:N2Y1MjAyNmIzMDEyZTBhODRjMTE3NGYxN2QyMDA4 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:N2YyZGJkZjQyYWJlMDlhNzcxNTA4NTE2NTBkNzA3 a=crypto:3 F8_128_HMAC_SHA1_80 inline:N2JmYTZkOGMxZDc4NmI5MDcwYWQ0OWEANWRmMGQx a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: --- (15 headers 14 lines) --- [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Sending to 10.200.158.208:5062 (no NAT) [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Using INVITE request as basis request - 433391560@10.200.158.208 [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Found peer '7309' for '7309' from 10.200.158.208:5062 [Feb 28 13:25:22] VERBOSE[29997] netsock2.c: == Using SIP RTP TOS bits 184 [Feb 28 13:25:22] VERBOSE[29997] netsock2.c: == Using SIP RTP CoS mark 5 [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Found RTP audio format 0 [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Found RTP audio format 8 [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Found RTP audio format 101 [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Found audio description format PCMU for ID 0 [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Found audio description format PCMA for ID 8 [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Found audio description format telephone-event for ID 101 [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Peer audio RTP is at port 10.200.158.208:11784 [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: Looking for 7303 in LOCAL (domain 10.200.104.112:5061) [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: list_route: hop: [Feb 28 13:25:22] VERBOSE[29997] chan_sip.c: <--- Transmitting (no NAT) to 10.200.158.208:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.200.158.208:5062;branch=z9hG4bK846448099;received=10.200.158.208 From: "7309@FLATESTE" ;tag=1080438895 To: Call-ID: 433391560@10.200.158.208 CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [7303@LOCAL:1] Goto("SIP/7309-00000012", "LOCAL_rulematch,7303,1") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Goto (LOCAL_rulematch,7303,1) [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [7303@LOCAL_rulematch:1] Macro("SIP/7309-00000012", "exten-vm,novm,7303") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/7309-00000012", "user-callerid") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/7309-00000012", "AMPUSER=7309") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/7309-00000012", "0?report") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/7309-00000012", "1?Set(REALCALLERIDNUM=7309)") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/7309-00000012", "AMPUSER=7309") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/7309-00000012", "AMPUSERCIDNAME=7309") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/7309-00000012", "0?report") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:7] Set("SIP/7309-00000012", "AMPUSERCID=7309") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:8] Set("SIP/7309-00000012", "CALLERID(all)="7309" <7309>") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/7309-00000012", "0?Set(CHANNEL(language)=)") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/7309-00000012", "0?continue") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:11] Set("SIP/7309-00000012", "__TTL=64") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:12] GotoIf("SIP/7309-00000012", "1?continue") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Goto (macro-user-callerid,s,19) [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/7309-00000012", "Using CallerID "7309" <7309>") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-exten-vm:2] Set("SIP/7309-00000012", "RingGroupMethod=none") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-exten-vm:3] Set("SIP/7309-00000012", "VMBOX=novm") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-exten-vm:4] Set("SIP/7309-00000012", "EXTTOCALL=7303") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/7309-00000012", "CFUEXT=") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-exten-vm:6] Set("SIP/7309-00000012", "CFBEXT=") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-exten-vm:7] Set("SIP/7309-00000012", "RT=""") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/7309-00000012", "record-enable,7303,IN") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/7309-00000012", "1?check") in new stack [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Goto (macro-record-enable,s,4) [Feb 28 13:25:22] VERBOSE[30262] pbx.c: -- Executing [s@macro-record-enable:4] AGI("SIP/7309-00000012", "recordingcheck,20110228-132522,1298910322.25") in new stack [Feb 28 13:25:22] VERBOSE[30262] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: recordingcheck,20110228-132522,1298910322.25: Inbound recording not enabled [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: -- AGI Script recordingcheck completed, returning 0 [Feb 28 13:25:23] VERBOSE[30262] pbx.c: -- Executing [s@macro-record-enable:5] MacroExit("SIP/7309-00000012", "") in new stack [Feb 28 13:25:23] VERBOSE[30262] pbx.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/7309-00000012", "dial,"",,7303") in new stack [Feb 28 13:25:23] VERBOSE[30262] pbx.c: -- Executing [s@macro-dial:1] GotoIf("SIP/7309-00000012", "1?dial") in new stack [Feb 28 13:25:23] VERBOSE[30262] pbx.c: -- Goto (macro-dial,s,3) [Feb 28 13:25:23] VERBOSE[30262] pbx.c: -- Executing [s@macro-dial:3] Set("SIP/7309-00000012", "_SIPSRTP_CRYPTO=268435456") in new stack [Feb 28 13:25:23] VERBOSE[30262] pbx.c: -- Executing [s@macro-dial:4] AGI("SIP/7309-00000012", "dialparties.agi") in new stack [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: dialparties.agi: Starting New Dialparties.agi [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: dialparties.agi: Caller ID name is '7309' number is '7309' [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: dialparties.agi: Methodology of ring is 'none' [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: -- dialparties.agi: Added extension 7303 to extension map [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: -- dialparties.agi: Extension 7303 cf is disabled [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: -- dialparties.agi: Extension 7303 do not disturb is disabled [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE) [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: -- dialparties.agi: dbset CALLTRACE/7303 to 7309 [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: -- dialparties.agi: Filtered ARG3: 7303 [Feb 28 13:25:23] VERBOSE[30262] res_agi.c: -- AGI Script dialparties.agi completed, returning 0 [Feb 28 13:25:23] VERBOSE[30262] pbx.c: -- Executing [s@macro-dial:8] Dial("SIP/7309-00000012", "SIP/7303,"",") in new stack [Feb 28 13:25:23] VERBOSE[30262] netsock2.c: == Using SIP RTP TOS bits 184 [Feb 28 13:25:23] VERBOSE[30262] netsock2.c: == Using SIP RTP CoS mark 5 [Feb 28 13:25:23] VERBOSE[30262] chan_sip.c: Audio is at 5061 [Feb 28 13:25:23] VERBOSE[30262] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Feb 28 13:25:23] VERBOSE[30262] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Feb 28 13:25:23] VERBOSE[30262] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 28 13:25:23] VERBOSE[30262] chan_sip.c: Reliably Transmitting (no NAT) to 10.200.104.104:5062: INVITE sip:7303@10.200.104.104:5062;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 10.200.104.112:5061;branch=z9hG4bK7eb457cd Max-Forwards: 70 From: "7309" ;tag=as74d82399 To: Contact: Call-ID: 413e48a2274ac9646cb271e21d879e60@10.200.104.112:5061 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.2.4 Date: Mon, 28 Feb 2011 16:25:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 347 v=0 o=root 547563778 547563778 IN IP4 10.200.104.112 s=Asterisk PBX 1.8.2.4 c=IN IP4 10.200.104.112 t=0 0 m=audio 11068 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:1CKTEp9p1+w6EXVlT2yWeyML4D9c7/qmJcaMBXbz --- [Feb 28 13:25:23] VERBOSE[30262] app_dial.c: -- Called 7303 [Feb 28 13:25:23] VERBOSE[30219] chan_sip.c: <--- SIP read from TLS:10.200.104.104:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.200.104.112:5061;branch=z9hG4bK7eb457cd From: "7309" ;tag=as74d82399 To: Call-ID: 413e48a2274ac9646cb271e21d879e60@10.200.104.112:5061 CSeq: 102 INVITE User-Agent: Yealink SIP-T22P 7.60.0.60 Content-Length: 0 <-------------> [Feb 28 13:25:23] VERBOSE[30219] chan_sip.c: --- (8 headers 0 lines) --- [Feb 28 13:25:24] VERBOSE[30219] chan_sip.c: <--- SIP read from TLS:10.200.104.104:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 10.200.104.112:5061;branch=z9hG4bK7eb457cd From: "7309" ;tag=as74d82399 To: ;tag=206803691 Call-ID: 413e48a2274ac9646cb271e21d879e60@10.200.104.112:5061 CSeq: 102 INVITE Contact: Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T22P 7.60.0.60 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 0 <-------------> [Feb 28 13:25:24] VERBOSE[30219] chan_sip.c: --- (11 headers 0 lines) --- [Feb 28 13:25:24] VERBOSE[30262] app_dial.c: -- SIP/7303-00000013 is ringing [Feb 28 13:25:24] VERBOSE[30262] chan_sip.c: <--- Transmitting (no NAT) to 10.200.158.208:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TLS 10.200.158.208:5062;branch=z9hG4bK846448099;received=10.200.158.208 From: "7309@FLATESTE" ;tag=1080438895 To: ;tag=as2435a950 Call-ID: 433391560@10.200.158.208 CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 28 13:25:24] VERBOSE[30219] chan_sip.c: <--- SIP read from TLS:10.200.104.104:5062 ---> <-------------> [Feb 28 13:25:27] VERBOSE[23586] asterisk.c: -- Remote UNIX connection [Feb 28 13:25:27] VERBOSE[30269] asterisk.c: -- Remote UNIX connection disconnected [Feb 28 13:25:29] VERBOSE[23621] chan_sip.c: Reliably Transmitting (no NAT) to 10.140.21.8:5060: OPTIONS sip:10.140.21.8 SIP/2.0 Via: SIP/2.0/UDP 10.200.104.112:5060;branch=z9hG4bK4c518676 Max-Forwards: 70 From: "Unknown" ;tag=as7c93a21b To: Contact: Call-ID: 0cf4ee726189ebaf7c14339712097a6e@10.200.104.112:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.4 Date: Mon, 28 Feb 2011 16:25:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 28 13:25:29] VERBOSE[23621] chan_sip.c: Reliably Transmitting (no NAT) to 10.140.27.8:5060: OPTIONS sip:10.140.27.8 SIP/2.0 Via: SIP/2.0/UDP 10.200.104.112:5060;branch=z9hG4bK4b21c913 Max-Forwards: 70 From: "Unknown" ;tag=as2e34192c To: Contact: Call-ID: 5defc5f52a00f89d67bf9d335db96223@10.200.104.112:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.4 Date: Mon, 28 Feb 2011 16:25:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 28 13:25:29] VERBOSE[23621] chan_sip.c: Reliably Transmitting (no NAT) to 10.140.79.8:5060: OPTIONS sip:10.140.79.8 SIP/2.0 Via: SIP/2.0/UDP 10.200.104.112:5060;branch=z9hG4bK74a0ab32 Max-Forwards: 70 From: "Unknown" ;tag=as26dc17a9 To: Contact: Call-ID: 7aae2856263906c64bfc2c25337e2cb5@10.200.104.112:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.4 Date: Mon, 28 Feb 2011 16:25:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 28 13:25:29] VERBOSE[23621] chan_sip.c: Reliably Transmitting (no NAT) to 10.140.51.8:5060: OPTIONS sip:10.140.51.8 SIP/2.0 Via: SIP/2.0/UDP 10.200.104.112:5060;branch=z9hG4bK381f07e4 Max-Forwards: 70 From: "Unknown" ;tag=as0ceac19e To: Contact: Call-ID: 09c5263833536a6934c7148b588985d7@10.200.104.112:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.4 Date: Mon, 28 Feb 2011 16:25:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 28 13:25:29] VERBOSE[23621] chan_sip.c: Reliably Transmitting (no NAT) to 10.140.65.8:5060: OPTIONS sip:10.140.65.8 SIP/2.0 Via: SIP/2.0/UDP 10.200.104.112:5060;branch=z9hG4bK40ee15c2 Max-Forwards: 70 From: "Unknown" ;tag=as4ddcce2a To: Contact: Call-ID: 181fd8287b36ba98522e05da6c52e93b@10.200.104.112:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.4 Date: Mon, 28 Feb 2011 16:25:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 28 13:25:29] VERBOSE[23621] chan_sip.c: Reliably Transmitting (no NAT) to 10.140.67.8:5060: OPTIONS sip:10.140.67.8 SIP/2.0 Via: SIP/2.0/UDP 10.200.104.112:5060;branch=z9hG4bK61919ac4 Max-Forwards: 70 From: "Unknown" ;tag=as2a3f3cf2 To: Contact: Call-ID: 1926e39d7945b0467527b318157282cf@10.200.104.112:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.4 Date: Mon, 28 Feb 2011 16:25:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 28 13:25:29] VERBOSE[23621] chan_sip.c: Reliably Transmitting (no NAT) to 10.140.63.8:5060: OPTIONS sip:10.140.63.8 SIP/2.0 Via: SIP/2.0/UDP 10.200.104.112:5060;branch=z9hG4bK242d0545 Max-Forwards: 70 From: "Unknown" ;tag=as39cce621 To: Contact: Call-ID: 3e3b8be45db9855e73f9602f1a326ec6@10.200.104.112:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.4 Date: Mon, 28 Feb 2011 16:25:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 28 13:25:29] VERBOSE[23621] chan_sip.c: Reliably Transmitting (no NAT) to 10.140.62.8:5060: OPTIONS sip:10.140.62.8 SIP/2.0 Via: SIP/2.0/UDP 10.200.104.112:5060;branch=z9hG4bK44aefff6 Max-Forwards: 70 From: "Unknown" ;tag=as1f723ba3 To: Contact: Call-ID: 5419d6ee79966b82147a4c964c22d697@10.200.104.112:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.4 Date: Mon, 28 Feb 2011 16:25:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: <--- SIP read from TLS:10.200.104.104:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.200.104.112:5061;branch=z9hG4bK7eb457cd From: "7309" ;tag=as74d82399 To: ;tag=206803691 Call-ID: 413e48a2274ac9646cb271e21d879e60@10.200.104.112:5061 CSeq: 102 INVITE Contact: Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T22P 7.60.0.60 Require: timer Supported: timer Min-SE: 90 Session-Expires: 3600;refresher=uac Content-Length: 288 v=0 o=- 20000 20000 IN IP4 10.200.104.104 s=SDP data c=IN IP4 10.200.104.104 t=0 0 m=audio 11780 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NTEzODZjYjUyNWQzZWY2NzEzNDhjYThlNmM1Njcy a=sendrecv a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 <-------------> [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: --- (15 headers 11 lines) --- [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: Found RTP audio format 8 [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: Found RTP audio format 101 [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: Found audio description format PCMA for ID 8 [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: Found audio description format telephone-event for ID 101 [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: Peer audio RTP is at port 10.200.104.104:11780 [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: list_route: hop: [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: set_destination: Parsing for address/port to send to [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: set_destination: set destination to 10.200.104.104:5062 [Feb 28 13:25:29] VERBOSE[30219] chan_sip.c: Transmitting (no NAT) to 10.200.104.104:5062: ACK sip:7303@10.200.104.104:5062;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 10.200.104.112:5061;branch=z9hG4bK685a1551 Max-Forwards: 70 From: "7309" ;tag=as74d82399 To: ;tag=206803691 Contact: Call-ID: 413e48a2274ac9646cb271e21d879e60@10.200.104.112:5061 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.4 Content-Length: 0 --- [Feb 28 13:25:29] VERBOSE[30262] app_dial.c: -- SIP/7303-00000013 answered SIP/7309-00000012 [Feb 28 13:25:29] VERBOSE[30262] chan_sip.c: Audio is at 5061 [Feb 28 13:25:29] VERBOSE[30262] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Feb 28 13:25:29] VERBOSE[30262] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Feb 28 13:25:29] VERBOSE[30262] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 28 13:25:29] VERBOSE[30262] chan_sip.c: <--- Reliably Transmitting (no NAT) to 10.200.158.208:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.200.158.208:5062;branch=z9hG4bK846448099;received=10.200.158.208 From: "7309@FLATESTE" ;tag=1080438895 To: ;tag=as2435a950 Call-ID: 433391560@10.200.158.208 CSeq: 2 INVITE Server: Asterisk PBX 1.8.2.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 349 v=0 o=root 1354642492 1354642492 IN IP4 10.200.104.112 s=Asterisk PBX 1.8.2.4 c=IN IP4 10.200.104.112 t=0 0 m=audio 14610 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:rQtSF/bhanmvK9oPxLtFyevK6+ebMvUD/K4FxKuv <------------> [Feb 28 13:25:30] VERBOSE[29997] chan_sip.c: <--- SIP read from TLS:10.200.158.208:5062 ---> ACK sip:7303@10.200.104.112:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 10.200.158.208:5062;branch=z9hG4bK1016463769 From: "7309@FLATESTE" ;tag=1080438895 To: ;tag=as2435a950 Call-ID: 433391560@10.200.158.208 CSeq: 2 ACK Contact: Max-Forwards: 70 User-Agent: Yealink SIP-T28P 2.60.0.80 Content-Length: 0 <-------------> [Feb 28 13:25:30] VERBOSE[29997] chan_sip.c: --- (10 headers 0 lines) --- [Feb 28 13:25:30] VERBOSE[30219] chan_sip.c: <--- SIP read from TLS:10.200.104.104:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS 10.200.104.112:5061;branch=z9hG4bK7eb457cd From: "7309" ;tag=as74d82399 To: ;tag=206803691 Call-ID: 413e48a2274ac9646cb271e21d879e60@10.200.104.112:5061 CSeq: 102 INVITE Contact: Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE User-Agent: Yealink SIP-T22P 7.60.0.60 Require: timer Supported: timer Min-SE: 90 Session-Expires: 3600;refresher=uac Content-Length: 288 v=0 o=- 20000 20000 IN IP4 10.200.104.104 s=SDP data c=IN IP4 10.200.104.104 t=0 0 m=audio 11780 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NTEzODZjYjUyNWQzZWY2NzEzNDhjYThlNmM1Njcy a=sendrecv a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 <-------------> [Feb 28 13:25:30] VERBOSE[30219] chan_sip.c: --- (15 headers 11 lines) --- [Feb 28 13:25:30] VERBOSE[30219] chan_sip.c: set_destination: Parsing for address/port to send to [Feb 28 13:25:30] VERBOSE[30219] chan_sip.c: set_destination: set destination to 10.200.104.104:5062 [Feb 28 13:25:30] VERBOSE[30219] chan_sip.c: Transmitting (no NAT) to 10.200.104.104:5062: ACK sip:7303@10.200.104.104:5062;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 10.200.104.112:5061;branch=z9hG4bK58e6b02a Max-Forwards: 70 From: "7309" ;tag=as74d82399 To: ;tag=206803691 Contact: Call-ID: 413e48a2274ac9646cb271e21d879e60@10.200.104.112:5061 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.2.4 Content-Length: 0