ast1*CLI> sip set debug peer 172.16.64.5
SIP Debugging Enabled for IP: 172.16.64.5:5060
    -- Executing [7469@h323:1] Dial("OOH323/myuser1-3", "SIP/7469@172.16.64.5") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 172.16.64.2 port 12946
Adding codec 0xffff (unknown) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.64.5:5060:
INVITE sip:7469@172.16.64.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.64.2:5060;branch=z9hG4bK3994780c;rport
Max-Forwards: 70
From: "root" <sip:8888@172.16.64.5>;tag=as37973b31
To: <sip:7469@172.16.64.5>
Contact: <sip:8888@172.16.64.2>
Call-ID: 3585bdb75432514c75310bad3efd3f76@172.16.64.5
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Tue, 16 Jun 2009 08:55:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 225

v=0
o=root 1949780791 1949780791 IN IP4 172.16.64.2
s=Asterisk PBX 1.6.1.1
c=IN IP4 172.16.64.2
t=0 0
m=audio 12946 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
    -- Called 7469@172.16.64.5
ast1*CLI> 
<--- SIP read from UDP://172.16.64.5:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 172.16.64.2:5060;branch=z9hG4bK3994780c;rport=5060
From: "root" <sip:8888@172.16.64.5>;tag=as37973b31
To: <sip:7469@172.16.64.5>
Call-ID: 3585bdb75432514c75310bad3efd3f76@172.16.64.5
CSeq: 102 INVITE
Server: OpenSIPS (1.5.1-notls (x86_64/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
ast1*CLI> 
<--- SIP read from UDP://172.16.64.5:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.16.64.2:5060;received=172.16.64.2;branch=z9hG4bK3994780c;rport=5060
From: "root" <sip:8888@172.16.64.5>;tag=as37973b31
To: <sip:7469@172.16.64.5>;tag=001906af068d000af6009469-608cbe3f
Call-ID: 3585bdb75432514c75310bad3efd3f76@172.16.64.5
Date: Tue, 16 Jun 2009 08:55:27 GMT
CSeq: 102 INVITE
Warning: 305 "Incompatible Media Formats"
Server: Cisco-CP7911G/8.5.2
Contact: <sip:7469@172.16.65.1:5060;transport=udp>
Record-Route: <sip:172.16.64.5;lr=on>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "7469" <sip:7469@172.16.64.5>;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0


<------------->
--- (15 headers 0 lines) ---
Transmitting (no NAT) to 172.16.64.5:5060:
ACK sip:7469@172.16.64.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.64.2:5060;branch=z9hG4bK3994780c;rport
Max-Forwards: 70
From: "root" <sip:8888@172.16.64.5>;tag=as37973b31
To: <sip:7469@172.16.64.5>;tag=001906af068d000af6009469-608cbe3f
Contact: <sip:8888@172.16.64.2>
Call-ID: 3585bdb75432514c75310bad3efd3f76@172.16.64.5
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0


---
    -- SIP/172.16.64.5-025d0b98 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [7469@h323:2] Dial("OOH323/myuser1-3", "SIP/7469@172.16.64.6") in new stack
  == Using SIP RTP CoS mark 5
Really destroying SIP dialog '3585bdb75432514c75310bad3efd3f76@172.16.64.5' Method: INVITE
    -- Called 7469@172.16.64.6
    -- SIP/172.16.64.6-02657528 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [7469@h323:3] Hangup("OOH323/myuser1-3", "") in new stack
  == Spawn extension (h323, 7469, 3) exited non-zero on 'OOH323/myuser1-3'