vox01*CLI> <-- SIP read from 192.117.233.146:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.117.233.157:5060;branch=z9hG4bK5f2a2115;rport;received=192.117.233.157 From: "1" ;tag=as76993143 To: ;tag=as7b8fb487 Call-ID: 1d18c9774896002d64381cba4dd5fcc4@192.117.233.157 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 189 =0x01*CLI> o=root 11780 11780 IN IP4 192.117.233.146 s=session c=IN IP4 192.117.233.146 t=0 0 m=audio 14022 RTP/AVP 8 3 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - --- (11 headers 9 lines)--- Found RTP audio format 8 Found RTP audio format 3 Peer audio RTP is at port 192.117.233.146:14022 Found description format PCMA Found description format GSM Capabilities: us - 0xa (gsm|alaw), peer - audio=0xa (gsm|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.117.233.146, port 5060 Transmitting (no NAT) to 192.117.233.146:5060: ACK sip:1234@192.117.233.146 SIP/2.0 Via: SIP/2.0/UDP 192.117.233.157:5060;branch=z9hG4bK40ff1a22;rport From: "1" ;tag=as76993143 To: ;tag=as7b8fb487 Contact: Call-ID: 1d18c9774896002d64381cba4dd5fcc4@192.117.233.157 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Nov 24 12:13:27 WARNING[5142]: channel.c:784 channel_find_locked: Avoided initial deadlock for '0x81b8d08', 10 retries! == Manager 'dialer' logged off from 127.0.0.1 vox01*CLI> sip no debug <-- SIP read from 192.117.233.146:5060: BYE sip:1@192.117.233.157 SIP/2.0 Via: SIP/2.0/UDP 192.117.233.146:5060;branch=z9hG4bK1b314c3f;rport From: ;tag=as7b8fb487 To: "1" ;tag=as76993143 Contact: Call-ID: 1d18c9774896002d64381cba4dd5fcc4@192.117.233.157 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 --- (9 headers 0 lines)--- Sending to 192.117.233.146 : 5060 (non-NAT) Transmitting (no NAT) to 192.117.233.146:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.117.233.146:5060;branch=z9hG4bK1b314c3f;rport;received=192.117.233.146 From: ;tag=as7b8fb487 To: "1" ;tag=as76993143 Call-ID: 1d18c9774896002d64381cba4dd5fcc4@192.117.233.157 CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Nov 24 12:13:57 WARNING[13651]: file.c:583 ast_readaudio_callback: Failed to write frame == Spawn extension (loader, s, 6) exited non-zero on 'SIP/teleconcept-9ae3' Destroying call '1d18c9774896002d64381cba4dd5fcc4@192.117.233.157' 12 headers, 0 linesebug Reliably Transmitting (no NAT) to 192.117.233.146:5060: OPTIONS sip:192.117.233.146 SIP/2.0 Via: SIP/2.0/UDP 192.117.233.157:5060;branch=z9hG4bK7e89bd78;rport From: "asterisk" ;tag=as54e0fbc9 To: Contact: Call-ID: 76217891578d2ade55f7f4ac6f3469f8@192.117.233.157 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 24 Nov 2005 12:13:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 linesebug Reliably Transmitting (no NAT) to 192.114.69.6:5060: OPTIONS sip:192.114.69.6 SIP/2.0 Via: SIP/2.0/UDP 192.117.233.157:5060;branch=z9hG4bK7aa97128;rport From: "asterisk" ;tag=as7de1abe9 To: Contact: Call-ID: 65f1abc972de6a066d74a1b30c4ddf8f@192.117.233.157 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 24 Nov 2005 12:13:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 debug