asterisk

atomical*CLI> 
    -- Attempting call on SIP/18155551212@sip.inphonex.com for s@outgoing:1 (Retry 1)
We're at 64.34.205.41 port 13958
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 208.239.76.163:5060:
INVITE sip:18155551212@sip.inphonex.com SIP/2.0
Via: SIP/2.0/UDP 64.34.205.41:5060;branch=z9hG4bK0b0da4c1;rport
From: "asterisk" <sip:asterisk@inphonex.com>;tag=as6bd12ed4
To: <sip:18155551212@sip.inphonex.com>
Contact: <sip:asterisk@64.34.205.41>
Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Mar 2007 04:45:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 13591 13591 IN IP4 64.34.205.41
s=session
c=IN IP4 64.34.205.41
t=0 0
m=audio 13958 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
atomical*CLI> 
<-- SIP read from 208.239.76.163:5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.34.205.41:5060;received=64.34.205.41;branch=z9hG4bK0b0da4c1;rport=5060
From: "asterisk" <sip:asterisk@inphonex.com>;tag=as6bd12ed4
To: <sip:18155551212@sip.inphonex.com>
Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com
CSeq: 102 INVITE


--- (6 headers 0 lines) ---
atomical*CLI> 
<-- SIP read from 208.239.76.163:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.34.205.41:5060;received=64.34.205.41;branch=z9hG4bK0b0da4c1;rport=5060
From: "asterisk" <sip:asterisk@inphonex.com>;tag=as6bd12ed4
To: <sip:18155551212@sip.inphonex.com>;tag=SDbasfb99-as2eefa867
Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com
CSeq: 102 INVITE
User-Agent: IPX Softswitch 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:18155551212@208.239.76.163:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 27288 27288 IN IP4 208.239.76.163
s=session
c=IN IP4 208.239.76.163
t=0 0
m=audio 25758 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (11 headers 12 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 208.239.76.163:25758
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:18155551212@208.239.76.163:5060;transport=udp>
set_destination: Parsing <sip:18155551212@208.239.76.163:5060;transport=udp> for address/port to send to
set_destination: set destination to 208.239.76.163, port 5060
Transmitting (no NAT) to 208.239.76.163:5060:
ACK sip:18155551212@208.239.76.163:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 64.34.205.41:5060;branch=z9hG4bK493dc3a7;rport
From: "asterisk" <sip:asterisk@inphonex.com>;tag=as6bd12ed4
To: <sip:18155551212@sip.inphonex.com>;tag=SDbasfb99-as2eefa867
Contact: <sip:asterisk@64.34.205.41>
Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
       > Channel SIP/sip.inphonex.com-0818baa8 was answered.
    -- Executing Monitor("SIP/sip.inphonex.com-0818baa8", "wav|mysound") in new stack
    -- Executing Wait("SIP/sip.inphonex.com-0818baa8", "15") in new stack
atomical*CLI> 
<-- SIP read from 208.239.76.163:5060: 
BYE sip:asterisk@64.34.205.41 SIP/2.0
Via: SIP/2.0/UDP 208.239.76.163:5060;branch=z9hG4bK0gbb3b10fgc0qgs3v300sd00000j1.1
From: <sip:18155551212@sip.inphonex.com>;tag=SDbasfb99-as2eefa867
To: "asterisk" <sip:asterisk@inphonex.com>;tag=as6bd12ed4
Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com
CSeq: 102 BYE
User-Agent: IPX Softswitch 2.0
Max-Forwards: 69
Content-Length: 0


--- (9 headers 0 lines) ---
Sending to 208.239.76.163 : 5060 (non-NAT)
Transmitting (no NAT) to 208.239.76.163:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.239.76.163:5060;branch=z9hG4bK0gbb3b10fgc0qgs3v300sd00000j1.1;received=208.239.76.163
From: <sip:18155551212@sip.inphonex.com>;tag=SDbasfb99-as2eefa867
To: "asterisk" <sip:asterisk@inphonex.com>;tag=as6bd12ed4
Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:asterisk@64.34.205.41>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
  == Spawn extension (outgoing, s, 2) exited non-zero on 'SIP/sip.inphonex.com-0818baa8'
Mar 28 04:45:38 NOTICE[15687]: pbx_spool.c:279 attempt_thread: Call completed to SIP/18155551212@sip.inphonex.com
Destroying call '21afa60f1b21ecba21b590a428187d2e@inphonex.com'