rendered paste body -- 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" <sip:debrecen-verety-trunk@10.128.4.30>;tag=as167f17b3
To: <sip:8887@10.52.4.250>
Contact: <sip:debrecen-verety-trunk@10.128.4.30>
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" <sip:debrecen-verety-trunk@10.128.4.30>;tag=as167f17b3
To: <sip:8887@10.52.4.250>;tag=as26e578b1
Contact: <sip:debrecen-verety-trunk@10.128.4.30>
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" <sip:debrecen-verety-trunk@10.128.4.30>;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