-- Executing [0748881658@numberplan-custom-1:1] Dial("SIP/23003-08238cd8", "SIP/10.0.0.1/0748881658||T") in new stack
Audio is at 10.0.0.2 port 13350
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.1:5060:
INVITE sip:0748881658@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK2dd2b9f3;rport
From: "Znuff" <sip:23003@10.0.0.2>;tag=as683db931
To: <sip:0748881658@10.0.0.1>
Contact: <sip:23003@10.0.0.2>
Call-ID: 649e745a678596ae5f5d5b62300851ff@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 06 Mar 2007 03:49:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 492

v=0
o=root 24756 24756 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 13350 RTP/AVP 0 3 8 112 5 10 7 110 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 10.0.0.1/0748881658
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 100 Trying
From: "Znuff"<sip:23003@10.0.0.2>;tag=as683db931
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-16e13-59619ac-5411
Call-ID: 649e745a678596ae5f5d5b62300851ff@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK2dd2b9f3;rport
Contact: <sip:0748881658@10.0.0.1>
Content-Length:0


<------------->
--- (8 headers 0 lines) ---
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
SIP/2.0 415 Unsupported Media Type
From: "Znuff"<sip:23003@10.0.0.2>;tag=as683db931
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-16e13-59619ac-5411
Call-ID: 649e745a678596ae5f5d5b62300851ff@10.0.0.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.2:5060;received=10.0.0.2;branch=z9hG4bK2dd2b9f3;rport
Content-Length:0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 415 "Unsupported Media Type" back from 10.0.0.1
Transmitting (no NAT) to 10.0.0.1:5060:
ACK sip:0748881658@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK2dd2b9f3;rport
From: "Znuff" <sip:23003@10.0.0.2>;tag=as683db931
To: <sip:0748881658@10.0.0.1>;tag=a000001-13c4-16e13-59619ac-5411
Contact: <sip:23003@10.0.0.2>
Call-ID: 649e745a678596ae5f5d5b62300851ff@10.0.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [0748881658@numberplan-custom-1:2] Congestion("SIP/23003-08238cd8", "") in new stack
  == Spawn extension (numberplan-custom-1, 0748881658, 2) exited non-zero on 'SIP/23003-08238cd8'
Really destroying SIP dialog '649e745a678596ae5f5d5b62300851ff@10.0.0.2' Method: INVITE
voip*CLI>