
<-- SIP read from 66.23.129.253:5060:
INVITE sip:7078231743@192.168.0.5:5060 SIP/2.0
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0
Max-Forwards: 16
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
Contact: Anonymous <sip:69.30.45.229:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 64486073-997163996-2289747427-1019163781
h323-conf-id: 64486073-997163996-2289747427-1019163781
Content-Type: application/sdp
Content-Length: 205

v=0
o=Sippy 269898604 0 IN IP4 69.30.45.229
s=-
a=sendrecv
t=0 0
m=audio 41498 RTP/AVP 18 0 101
c=IN IP4 69.30.45.229
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

--- (15 headers 10 lines) ---
Using INVITE request as basis request - 1179663072-3535348@LA4_SIP_01
Sending to 66.23.129.253 : 5060 (NAT)
Found peer '66.23.129.253'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 69.30.45.229:41498
Found description format G729
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 7078231743 in from-sip-external (domain 192.168.0.5)
list_route: hop: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
Transmitting (no NAT) to 66.23.129.253:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Length: 0


---
    -- Executing NoOp("SIP/mushrooms-b7a12120", "Received incoming SIP connection from unknown peer to 7078231743") in new stack
    -- Executing Set("SIP/mushrooms-b7a12120", "DID=7078231743") in new stack
    -- Executing Goto("SIP/mushrooms-b7a12120", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing Ringing("SIP/mushrooms-b7a12120", "") in new stack
Transmitting (no NAT) to 66.23.129.253:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Length: 0


---
    -- Executing GotoIf("SIP/mushrooms-b7a12120", "1?from-trunk|7078231743|1") in new stack
    -- Goto (from-trunk,7078231743,1)
    -- Executing Set("SIP/mushrooms-b7a12120", "FROM_DID=7078231743") in new stack
    -- Executing Gosub("SIP/mushrooms-b7a12120", "app-blacklist-check|s|1") in new stack
    -- Executing LookupBlacklist("SIP/mushrooms-b7a12120", "") in new stack
    -- Executing GotoIf("SIP/mushrooms-b7a12120", "0?blacklisted") in new stack
    -- Executing Return("SIP/mushrooms-b7a12120", "") in new stack
    -- Executing Goto("SIP/mushrooms-b7a12120", "ivr-2|s|1") in new stack
    -- Goto (ivr-2,s,1)
    -- Executing Set("SIP/mushrooms-b7a12120", "LOOPCOUNT=0") in new stack
    -- Executing Set("SIP/mushrooms-b7a12120", "__DIR-CONTEXT=default") in new stack
    -- Executing Set("SIP/mushrooms-b7a12120", "_IVR_CONTEXT_ivr-2=") in new stack
    -- Executing Set("SIP/mushrooms-b7a12120", "_IVR_CONTEXT=ivr-2") in new stack
    -- Executing GotoIf("SIP/mushrooms-b7a12120", "0?begin") in new stack
    -- Executing Answer("SIP/mushrooms-b7a12120", "") in new stack
We're at 74.95.5.154 port 14228
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 66.23.129.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 10040 10040 IN IP4 74.95.5.154
s=session
c=IN IP4 74.95.5.154
t=0 0
m=audio 14228 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Executing Wait("SIP/mushrooms-b7a12120", "1") in new stack
trixbox*CLI>
<-- SIP read from 66.23.129.253:5060:
INVITE sip:7078231743@192.168.0.5:5060 SIP/2.0
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0
Max-Forwards: 16
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
Contact: Anonymous <sip:69.30.45.229:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 64486073-997163996-2289747427-1019163781
h323-conf-id: 64486073-997163996-2289747427-1019163781
Content-Type: application/sdp
Content-Length: 205

v=0
o=Sippy 269898604 0 IN IP4 69.30.45.229
s=-
a=sendrecv
t=0 0
m=audio 41498 RTP/AVP 18 0 101
c=IN IP4 69.30.45.229
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

--- (15 headers 10 lines) ---
Ignoring this INVITE request
We're at 74.95.5.154 port 14228
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 66.23.129.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 10040 10041 IN IP4 74.95.5.154
s=session
c=IN IP4 74.95.5.154
t=0 0
m=audio 14228 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
trixbox*CLI>
<-- SIP read from 66.23.129.253:5060:
ACK sip:7078231743@192.168.0.5:5060 SIP/2.0
Record-Route: <sip:7078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=0
Max-Forwards: 16
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 ACK
Expires: 300
User-Agent: Sippy
Content-Length: 0


--- (11 headers 0 lines) ---
Retransmitting #1 (no NAT) to 66.23.129.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 10040 10040 IN IP4 74.95.5.154
s=session
c=IN IP4 74.95.5.154
t=0 0
m=audio 14228 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Executing Set("SIP/mushrooms-b7a12120", "TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3
    -- Executing Set("SIP/mushrooms-b7a12120", "TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing BackGround("SIP/mushrooms-b7a12120", "custom/NewMain") in new stack
    -- Playing 'custom/NewMain' (language 'en')
Retransmitting #2 (no NAT) to 66.23.129.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 10040 10040 IN IP4 74.95.5.154
s=session
c=IN IP4 74.95.5.154
t=0 0
m=audio 14228 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Retransmitting #3 (no NAT) to 66.23.129.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 10040 10040 IN IP4 74.95.5.154
s=session
c=IN IP4 74.95.5.154
t=0 0
m=audio 14228 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Retransmitting #4 (no NAT) to 66.23.129.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 10040 10040 IN IP4 74.95.5.154
s=session
c=IN IP4 74.95.5.154
t=0 0
m=audio 14228 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Retransmitting #5 (no NAT) to 66.23.129.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 10040 10040 IN IP4 74.95.5.154
s=session
c=IN IP4 74.95.5.154
t=0 0
m=audio 14228 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Retransmitting #6 (no NAT) to 66.23.129.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKc9a3.a287c4e5.0;received=66.23.129.253
Record-Route: <sip:17078231743@66.23.129.253:5060;lr;nat=yes;ftag=ce89ccddbc01911022553bdb99d28464>
From: <sip:4147632332@69.30.45.229>;tag=ce89ccddbc01911022553bdb99d28464
To: <sip:17078231743@66.23.129.253>;tag=as1b70877b
Call-ID: 1179663072-3535348@LA4_SIP_01
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7078231743@74.95.5.154>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 10040 10040 IN IP4 74.95.5.154
s=session
c=IN IP4 74.95.5.154
t=0 0
m=audio 14228 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
  == Spawn extension (ivr-2, s, 10) exited non-zero on 'SIP/mushrooms-b7a12120'
    -- Executing Hangup("SIP/mushrooms-b7a12120", "") in new stack
  == Spawn extension (ivr-2, h, 1) exited non-zero on 'SIP/mushrooms-b7a12120'
Destroying call '1179663072-3535348@LA4_SIP_01'










host=74.95.5.154