Unnamed

pbx*CLI>
<-- SIP read from Y.Y.Y.Y:6252:
INVITE sip:31101@pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.17:6252;branch=z9hG4bK-d87543-6e76826f93176173-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:joelaptop@192.168.22.17:6252>
To: <sip:31101@pbx.elephantoutlook.com>
From: "joe Webster <3149>"<sip:joelaptop@pbx.elephantoutlook.com>;tag=42335900
Call-ID: f7573803537aca4f@b2JzZXJ2ZXI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="joelaptop",realm="asterisk",nonce="58f0ff85",uri="sip:31101@pbx.elephantoutlook.com",response="417fc459dd52345ac058e8da5a23f5c5",algorithm=MD5
Supported: eventlist
User-Agent: eyeBeam release 3010n stamp 19039
Content-Length: 301

v=0
o=- 70358654 70358725 IN IP4 192.168.22.17
s=eyeBeam
c=IN IP4 192.168.22.17
t=0 0
m=audio 6254 RTP/AVP 100 6 0 8 3 18 97 5 101
a=alt:1 1 : 9B24FA22 0000001C 192.168.22.17 6254
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (14 headers 12 lines)---
Using INVITE request as basis request - f7573803537aca4f@b2JzZXJ2ZXI.
Sending to 192.168.22.17 : 6252 (NAT)
Found user 'joelaptop'
Found RTP audio format 100
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 192.168.22.17:6254
Peer video RTP is at port 192.168.22.17:65535
Found description format speex
Found description format speex
Found description format telephone-event
Capabilities: us - 0x80006 (gsm|ulaw|h263), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 31101 in local (domain pbx.elephantoutlook.com)
list_route: hop: <sip:joelaptop@192.168.22.17:6252>
Transmitting (NAT) to Y.Y.Y.Y:6252:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.22.17:6252;branch=z9hG4bK-d87543-6e76826f93176173-1--d87543-;received=Y.Y.Y.Y;rport=6252
From: "joe Webster <3149>"<sip:joelaptop@pbx.elephantoutlook.com>;tag=42335900
To: <sip:31101@pbx.elephantoutlook.com>
Call-ID: f7573803537aca4f@b2JzZXJ2ZXI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:31101@66.192.107.225>
Content-Length: 0


---
    -- Executing Dial("SIP/joelaptop-b6a095f0", "Local/moh@local||tT") in new stack
    -- Called moh@local
    -- Executing WaitMusicOnHold("Local/moh@local-df08,2", "300") in new stack
We're at 66.192.107.225 port 16142
Video is at 66.192.107.225 port 16434
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to Y.Y.Y.Y:6252:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.22.17:6252;branch=z9hG4bK-d87543-6e76826f93176173-1--d87543-;received=Y.Y.Y.Y;rport=6252
From: "joe Webster <3149>"<sip:joelaptop@pbx.elephantoutlook.com>;tag=42335900
To: <sip:31101@pbx.elephantoutlook.com>;tag=as1098c46b
Call-ID: f7573803537aca4f@b2JzZXJ2ZXI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:31101@66.192.107.225>
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 28675 28675 IN IP4 66.192.107.225
s=session
c=IN IP4 66.192.107.225
t=0 0
m=audio 16142 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Started music on hold, class 'default', on channel 'Local/moh@local-df08,2'
pbx*CLI>