-- Attempting call on SIP/voicepulse-primary/y9T2sFG96n for application SMS(0) (Retry 5)
  == Using SIP RTP CoS mark 5
    -- Got SIP response 500 "Nice try" back from 64.61.93.190
       > Channel SIP/voicepulse-primary-000020d3 was never answered.
[2012-12-28 16:46:42] NOTICE[31284]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)

- - - - - - - - - -

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '0ac51f68631d6fb16e16767b7969462a@75.144.202.33' Method: OPTIONS
    -- Attempting call on SIP/voicepulse-primary/y9T2sFG96n for application SMS(0) (Retry 2)
  == Using SIP RTP CoS mark 5
We think we can do text
Audio is at 75.144.202.33 port 11934
Adding codec 0x40 (slin) to SDP
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 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.61.93.190:5060:
INVITE sip:y9T2sFG96n@jfk-primary.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 75.144.202.33:5060;branch=z9hG4bK2cc62c9b;rport
Max-Forwards: 70
From: "SMS" <sip:0@75.144.202.33>;tag=as65d13ba3
To: <sip:y9T2sFG96n@jfk-primary.voicepulse.com>
Contact: <sip:0@75.144.202.33>
Call-ID: 7678a432511ad8157c41c397632d695a@75.144.202.33
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1.2
Date: Fri, 28 Dec 2012 21:39:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 488

v=0
o=root 2086454903 2086454903 IN IP4 75.144.202.33
s=Asterisk PBX 1.6.2.7-1ubuntu1.2
c=IN IP4 75.144.202.33
t=0 0
m=audio 11934 RTP/AVP 10 0 3 8 112 5 7 110 111 9 101
a=rtpmap:10 L16/8000
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:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
LC-phonenew*CLI>
<--- SIP read from UDP:64.61.93.190:5060 --->
SIP/2.0 500 Nice try
Via: SIP/2.0/UDP 75.144.202.33:5060;branch=z9hG4bK2cc62c9b;rport=5060
From: "SMS" <sip:0@75.144.202.33>;tag=as65d13ba3
To: <sip:y9T2sFG96n@jfk-primary.voicepulse.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e641
Call-ID: 7678a432511ad8157c41c397632d695a@75.144.202.33
CSeq: 102 INVITE
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 500 "Nice try" back from 64.61.93.190
Transmitting (no NAT) to 64.61.93.190:5060:
ACK sip:y9T2sFG96n@jfk-primary.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 75.144.202.33:5060;branch=z9hG4bK2cc62c9b;rport
Max-Forwards: 70
From: "SMS" <sip:0@75.144.202.33>;tag=as65d13ba3
To: <sip:y9T2sFG96n@jfk-primary.voicepulse.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.e641
Contact: <sip:0@75.144.202.33>
Call-ID: 7678a432511ad8157c41c397632d695a@75.144.202.33
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1.2
Content-Length: 0

---
       > Channel SIP/voicepulse-primary-000020c6 was never answered.
[2012-12-28 16:39:57] NOTICE[31248]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)
Really destroying SIP dialog '3efb1179-8e311880-e4e4da8b@192.168.0.252' Method: REGISTER
Really destroying SIP dialog '7678a432511ad8157c41c397632d695a@75.144.202.33' Method: INVITE
LC-phonenew*CLI>
<--- SIP read from UDP:71.192.46.170:34099 --->

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