--- SIP read from 8.8.194.3:5065 ---> INVITE sip:7119765421@204.126.120.201:5060 SIP/2.0 From: "Asterisk TestPhone" ;tag=8af60d5c-1dd2-11b2-bcc4-b03162323164+8afe005c Contact: "Asterisk TestPhone" To: Call-ID: 59998-1944843129@8.8.194.3 CSeq: 1 INVITE Via: SIP/2.0/UDP 8.8.194.3:5065;branch=z9hG4bK1913608549512361 Max-Forwards: 70 Allow: INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY Content-Type: application/sdp Content-Length: 195 v=0 o=- 3381628222 3381628222 IN IP4 8.8.194.3 s=MGCP Call c=IN IP4 65.89.123.10 t=0 0 m=audio 16704 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (11 headers 9 lines) --- Sending to 8.8.194.3 : 5065 (no NAT) Using INVITE request as basis request - 59998-1944843129@8.8.194.3 [Dec 11 09:15:36] DEBUG[6132]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Dec 11 09:15:36] DEBUG[6132]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '5026382680' Found user '5026382680' Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 65.89.123.10:16704 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 65.89.123.10:16704 Looking for 7119765421 in lyvmail-test (domain 204.126.120.201) list_route: hop: <--- Transmitting (no NAT) to 8.8.194.3:5065 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 8.8.194.3:5065;branch=z9hG4bK1913608549512361;received=8.8.194.3 From: "Asterisk TestPhone" ;tag=8af60d5c-1dd2-11b2-bcc4-b03162323164+8afe005c To: Call-ID: 59998-1944843129@8.8.194.3 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 11 09:15:36] DEBUG[6110]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Dec 11 09:15:36] DEBUG[6110]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '5026382680' -- Executing [7119765421@lyvmail-test:1] Goto("SIP/5026382680-007f5840", "lyvmail-test-run|s|1") in new stack -- Goto (lyvmail-test-run,s,1) -- Executing [s@lyvmail-test-run:1] Answer("SIP/5026382680-007f5840", "") in new stack Audio is at 204.126.120.201 port 14302 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP localhost*CLI> <--- Reliably Transmitting (no NAT) to 8.8.194.3:5065 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 8.8.194.3:5065;branch=z9hG4bK1913608549512361;received=8.8.194.3 From: "Asterisk TestPhone" ;tag=8af60d5c-1dd2-11b2-bcc4-b03162323164+8afe005c To: ;tag=as3f0a9b38 Call-ID: 59998-1944843129@8.8.194.3 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=root 6030 6030 IN IP4 204.126.120.201 s=session c=IN IP4 204.126.120.201 t=0 0 m=audio 14302 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Executing [s@lyvmail-test-run:2] BackGround("SIP/5026382680-007f5840", "/var/lib/asterisk/sounds/mysounds/sounds/go-away2") in new stack -- Playing '/var/lib/asterisk/sounds/mysounds/sounds/go-away2' (language 'en') localhost*CLI>