--- (10 headers 0 lines)--- aries*CLI> <-- SIP read from 72.40.10.150:5060: INVITE sip:7777@72.232.182.13 SIP/2.0 Via: SIP/2.0/UDP 72.40.10.150:5060;branch=z9hG4bK-795593e6 From: 200 ;tag=3f75ecdf361b2c92o1 To: Call-ID: 879428a3-b7fa839e@72.40.10.150 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="41f6d136",uri="sip:7777@72.232.182.13",algorithm=MD5,response="d757cb8a17883c81fad24d2a1f8e673a" Contact: 200 xpires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 426 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 16357734 16357734 IN IP4 72.40.10.150 s=- c=IN IP4 72.40.10.150 t=0 0 m=audio 16426 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 19 lines)--- Using INVITE request as basis request - 879428a3-b7fa839e@72.40.10.150 Sending to 72.40.10.150 : 5060 (non-NAT) Found user '200' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 72.40.10.150:16426 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 7777 in internal (domain 72.232.182.13) list_route: hop: Transmitting (no NAT) to 72.40.10.150:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 72.40.10.150:5060;branch=z9hG4bK-795593e6;received=72.40.10.150 From: 200 ;tag=3f75ecdf361b2c92o1 To: Call-ID: 879428a3-b7fa839e@72.40.10.150 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 10 19:44:56 WARNING[14313]: pbx.c:2357 __ast_pbx_run: Channel 'SIP/200-088e' sent into invalid extension '8006170141' in context 'internal', but no invalid handler Jul 10 19:44:56 ERROR[14313]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql: cannot connect to database server localhost. Reliably Transmitting (no NAT) to 72.40.10.150:5060: SIP/2.0 603 Declined Via: SIP/2.0/UDP 72.40.10.150:5060;branch=z9hG4bK-795593e6;received=72.40.10.150 From: 200 ;tag=3f75ecdf361b2c92o1 To: ;tag=as46fa1e99 Call-ID: 879428a3-b7fa839e@72.40.10.150 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- aries*CLI> <-- SIP read from 72.40.10.150:5060: ACK sip:7777@72.232.182.13 SIP/2.0 Via: SIP/2.0/UDP 72.40.10.150:5060;branch=z9hG4bK-795593e6 From: 200 ;tag=3f75ecdf361b2c92o1 To: ;tag=as46fa1e99 Call-ID: 879428a3-b7fa839e@72.40.10.150 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="41f6d136",uri="sip:7777@72.232.182.13",algorithm=MD5,response="5344fd01f6f131c64b53a2cfc6f636f3" Contact: 200 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '879428a3-b7fa839e@72.40.10.150' aries*CLI>