--- Destroying call 'D4545174-1DD1-11B2-9DE2-8958C5C3E31D@192.168.1.177' backup*CLI> <-- SIP read from 192.168.1.20:5060: INVITE sip:403@192.168.1.127 SIP/2.0 Content-Length: 366 Contact: Call-ID: C0D68F86-1DD1-11B2-9227-862647F31DD3@192.168.1.20 Content-Type: application/sdp From: "Tony,,,";tag=564649524581758548 CSeq: 1 INVITE Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.1.20;rport;branch=z9hG4bKc0a801140131c9b14256709b0c7a546000000229 User-Agent: SJphone/1.50.273d (SJ Labs) v=0 o=- 3321949979 3321949979 IN IP4 62.168.4.1 s=SJphone c=IN IP4 62.168.4.1 t=0 0 a=direction:active m=audio 49172 RTP/AVP 3 97 98 110 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 16 lines)--- Using latest request as basis request backup*CLI> Sending to 192.168.1.20 : 5060 (non-NAT) backup*CLI> Reliably Transmitting (no NAT) to 192.168.1.20:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bKc0a801140131c9b14256709b0c7a546000000229 From: "Tony,,,";tag=564649524581758548 To: ;tag=as77d6e991 Call-ID: C0D68F86-1DD1-11B2-9227-862647F31DD3@192.168.1.20 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="0e742383" Content-Length: 0 --- backup*CLI> Scheduling destruction of call 'C0D68F86-1DD1-11B2-9227-862647F31DD3@192.168.1.20' in 15000 ms backup*CLI> Found user '204' backup*CLI> <-- SIP read from 192.168.1.20:5060: ACK sip:403@192.168.1.127 SIP/2.0 Content-Length: 0 Call-ID: C0D68F86-1DD1-11B2-9227-862647F31DD3@192.168.1.20 Max-Forwards: 70 CSeq: 1 ACK From: "Tony,,,";tag=564649524581758548 To: ;tag=as77d6e991 Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bKc0a801140131c9b14256709b0c7a546000000229 backup*CLI> --- (8 headers 0 lines)--- backup*CLI> <-- SIP read from 192.168.1.20:5060: INVITE sip:403@192.168.1.127 SIP/2.0 Content-Length: 366 Contact: Call-ID: C0D68F86-1DD1-11B2-9227-862647F31DD3@192.168.1.20 Content-Type: application/sdp From: "Tony,,,";tag=564649524581758548 CSeq: 2 INVITE Max-Forwards: 70 To: Via: SIP/2.0/UDP 192.168.1.20;rport;branch=z9hG4bKc0a80114000000be4256709b4cd1e99f0000022b User-Agent: SJphone/1.50.273d (SJ Labs) Proxy-Authorization: Digest username="204",realm="asterisk",nonce="0e742383",uri="sip:403@192.168.1.127",response="cb19335c9f42641e619a1b8ec1db556c" v=0 o=- 3321949979 3321949979 IN IP4 62.168.4.1 s=SJphone c=IN IP4 62.168.4.1 t=0 0 a=direction:active m=audio 49172 RTP/AVP 3 97 98 110 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 backup*CLI> --- (12 headers 16 lines)--- backup*CLI> Using latest request as basis request backup*CLI> Sending to 192.168.1.20 : 5060 (non-NAT) backup*CLI> Found user '204' backup*CLI> Found RTP audio format 3 backup*CLI> Found RTP audio format 97 backup*CLI> Found RTP audio format 98 backup*CLI> Found RTP audio format 110 backup*CLI> Found RTP audio format 8 backup*CLI> Found RTP audio format 0 backup*CLI> Found RTP audio format 101 backup*CLI> Peer audio RTP is at port 62.168.4.1:49172 backup*CLI> Found description format GSM backup*CLI> Found description format iLBC backup*CLI> Found description format iLBC backup*CLI> Found description format speex backup*CLI> Found description format PCMA backup*CLI> Found description format PCMU backup*CLI> Found description format telephone-event backup*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) backup*CLI> Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) backup*CLI> Looking for 403 in sip_phone backup*CLI> list_route: hop: backup*CLI> Transmitting (no NAT) to 192.168.1.20:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bKc0a80114000000be4256709b4cd1e99f0000022b From: "Tony,,,";tag=564649524581758548 To: Call-ID: C0D68F86-1DD1-11B2-9227-862647F31DD3@192.168.1.20 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- backup*CLI> -- Executing Dial("SIP/204-5fc9", "SIP/403") in new stack backup*CLI> Destroying call '5734b547653d1ba629bb051943502640@192.168.1.127' backup*CLI> Apr 8 13:52:55 NOTICE[10812]: app_dial.c:960 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) backup*CLI> == Everyone is busy/congested at this time (1:0/1/0) backup*CLI> == Auto fallthrough, channel 'SIP/204-5fc9' status is 'CONGESTION' backup*CLI> Transmitting (no NAT) to 192.168.1.20:5060: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bKc0a80114000000be4256709b4cd1e99f0000022b From: "Tony,,,";tag=564649524581758548 To: ;tag=as328e140d Call-ID: C0D68F86-1DD1-11B2-9227-862647F31DD3@192.168.1.20 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- backup*CLI> <-- SIP read from 192.168.1.20:5060: ACK sip:403@192.168.1.127 SIP/2.0 Content-Length: 0 Call-ID: C0D68F86-1DD1-11B2-9227-862647F31DD3@192.168.1.20 Max-Forwards: 70 CSeq: 2 ACK From: "Tony,,,";tag=564649524581758548 To: ;tag=as328e140d Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bKc0a80114000000be4256709b4cd1e99f0000022b backup*CLI> --- (8 headers 0 lines)--- backup*CLI> Destroying call 'C0D68F86-1DD1-11B2-9227-862647F31DD3@192.168.1.20' backup*CLI> 11 headers, 0 lines Reliably Transmitting (no NAT) to 62.168.4.1:5060: OPTIONS sip:403@62.168.4.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.127:5060;branch=z9hG4bK610deb2a From: "asterisk" ;tag=as40330c39 To: Contact: Call-ID: 04360ff04a6b4de87459c68f4c5d1bf8@192.168.1.127 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Fri, 08 Apr 2005 11:53:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 --- backup*CLI>