-- Executing [8887@default:1] Dial("SIP/5022-b6908f00", "SIP/debrecen-verety-trunk/8887") in new stack Audio is at 10.128.4.30 port 13362 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.52.4.250:5060: INVITE sip:8887@10.52.4.250 SIP/2.0 Via: SIP/2.0/UDP 10.128.4.30:5060;branch=z9hG4bK1d8121c5;rport From: "Nick Thompson" ;tag=as167f17b3 To: Contact: Call-ID: 49958c7f5499cbbc10f35c4256fcf5ca@10.128.4.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 28 Jun 2007 18:31:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 330 v=0 o=root 32220 32220 IN IP4 10.128.4.30 s=session c=IN IP4 10.128.4.30 t=0 0 m=audio 13362 RTP/AVP 0 8 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called debrecen-verety-trunk/8887 Transmitting (no NAT) to 10.52.4.250:5060: ACK sip:8887@10.52.4.250 SIP/2.0 Via: SIP/2.0/UDP 10.128.4.30:5060;branch=z9hG4bK1d8121c5;rport From: "Nick Thompson" ;tag=as167f17b3 To: ;tag=as26e578b1 Contact: Call-ID: 49958c7f5499cbbc10f35c4256fcf5ca@10.128.4.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jun 28 13:31:05] NOTICE[32251]: chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"Nick Thompson" ;tag=as167f17b3' -- SIP/debrecen-verety-trunk-0839cd20 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/5022-b6908f00' status is 'CONGESTION' Really destroying SIP dialog '49958c7f5499cbbc10f35c4256fcf5ca@10.128.4.30' Method: INVITE