<------------>
    -- Executing [99319804297@myphones:1] Log("SIP/100-0000000e", "NOTICE, Dialing Out from "DoghouseComputers" <100> to 9319804297 through simpsig") in new stack
[Mar 29 14:43:34] NOTICE[2298]: Ext. 99319804297:1 @ myphones:  Dialing Out from "DoghouseComputers" <100> to 9319804297 through simpsig
    -- Executing [99319804297@myphones:2] Dial("SIP/100-0000000e", "SIP/simpsig/9319804297") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 71.87.181.173 port 16360
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 208.77.200.13:5060:
INVITE sip:9319804297@trunk.myvtel.com SIP/2.0
Via: SIP/2.0/UDP 71.87.181.173:5060;branch=z9hG4bK13e5dbff;rport
Max-Forwards: 70
From: "DoghouseComputers" <sip:100@myvtel.com>;tag=as23f781c5
To: <sip:9319804297@trunk.myvtel.com>
Contact: <sip:100@71.87.181.173>
Call-ID: 35a711dd0161a38b557c642470f8a70a@myvtel.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Tue, 29 Mar 2011 19:43:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 329122670 329122670 IN IP4 71.87.181.173
s=Asterisk PBX 1.6.2.11
c=IN IP4 71.87.181.173
t=0 0
m=audio 16360 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called simpsig/9319804297

<--- SIP read from UDP:208.77.200.13:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 71.87.181.173:5060;branch=z9hG4bK13e5dbff;rport=5060
From: "DoghouseComputers" <sip:100@myvtel.com>;tag=as23f781c5
To: <sip:9319804297@trunk.myvtel.com>
Call-ID: 35a711dd0161a38b557c642470f8a70a@myvtel.com
CSeq: 102 INVITE


<------------->
--- (6 headers 0 lines) ---

<--- SIP read from UDP:208.77.200.13:5060 --->
SIP/2.0 604 Does not exist anywhere
Via: SIP/2.0/UDP 71.87.181.173:5060;branch=z9hG4bK13e5dbff;rport=5060
From: "DoghouseComputers" <sip:100@myvtel.com>;tag=as23f781c5
To: <sip:9319804297@trunk.myvtel.com>;tag=SDs7dg499-596081199-1301467599087
Call-ID: 35a711dd0161a38b557c642470f8a70a@myvtel.com
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 604 "Does not exist anywhere" back from 208.77.200.13
Transmitting (NAT) to 208.77.200.13:5060:
ACK sip:9319804297@trunk.myvtel.com SIP/2.0
Via: SIP/2.0/UDP 71.87.181.173:5060;branch=z9hG4bK13e5dbff;rport
Max-Forwards: 70
From: "DoghouseComputers" <sip:100@myvtel.com>;tag=as23f781c5
To: <sip:9319804297@trunk.myvtel.com>;tag=SDs7dg499-596081199-1301467599087
Contact: <sip:100@71.87.181.173>
Call-ID: 35a711dd0161a38b557c642470f8a70a@myvtel.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [99319804297@myphones:3] PlayTones("SIP/100-0000000e", "congestion") in new stack
    -- Executing [99319804297@myphones:4] Hangup("SIP/100-0000000e", "") in new stack
  == Spawn extension (myphones, 99319804297, 4) exited non-zero on 'SIP/100-0000000e'
Scheduling destruction of SIP dialog 'NzgxMzQ3ZDdjMzZmMzZlYmIyMjVlYTJlYzYwNGI3Nzg.' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.1.11:20728 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.11:20728;branch=z9hG4bK-d8754z-c1721527b7898848-1---d8754z-;received=192.168.1.11;rport=20728
From: "DoghouseComputers"<sip:100@dhctech.net>;tag=ca2a07f5
To: "99319804297"<sip:99319804297@dhctech.net>;tag=as7f3e3b99
Call-ID: NzgxMzQ3ZDdjMzZmMzZlYmIyMjVlYTJlYzYwNGI3Nzg.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.11:20728 --->
ACK sip:99319804297@dhctech.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:20728;branch=z9hG4bK-d8754z-c1721527b7898848-1---d8754z-;rport
Max-Forwards: 70
To: "99319804297"<sip:99319804297@dhctech.net>;tag=as7f3e3b99
From: "DoghouseComputers"<sip:100@dhctech.net>;tag=ca2a07f5
Call-ID: NzgxMzQ3ZDdjMzZmMzZlYmIyMjVlYTJlYzYwNGI3Nzg.
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '35a711dd0161a38b557c642470f8a70a@myvtel.com' Method: INVITE