
<--- SIP read from 211.126.32.246:5060 --->
INVITE sip:6787721321@10.212.7.226 SIP/2.0
Via: SIP/2.0/UDP 211.126.32.246:5060;branch=z9hG4bK9uedvc30eg4h7c0r43o0sbepner80.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info
Max-Forwards: 69
Call-ID: 1bcb90662f4a4e2379076fa35882452f@10.212.7.226
From: <sip:14044084026@metaswitch.phoneprovider.com>;tag=SDe3mh899-10.246.15.8+1+3da3a3+db287c2a
To: "Chris McGarty" <sip:6787721321@10.212.7.226>;tag=as55882f8a
CSeq: 778024721 INVITE
Expires: 180I>
Contact: <sip:14044084026@211.126.32.246:5060;transport=udp>
Supported: 100rel
Content-Length: 167
Content-Type: application/sdp

v=0
o=- 1554245977 1554245977 IN IP4 211.126.32.246
s=-
c=IN IP4 211.126.32.246
t=0 0
m=audio 14842 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
--- (13 headers 8 lines) ---
Sending to 211.126.32.246 : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 211.126.32.246:14842
Found audio description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 211.126.32.246:14842
localhost*CLI>
<--- Transmitting (no NAT) to 211.126.32.246:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 211.126.32.246:5060;branch=z9hG4bK9uedvc30eg4h7c0r43o0sbepner80.1;received=211.126.32.246
From: <sip:14044084026@metaswitch.phoneprovider.com>;tag=SDe3mh899-10.246.15.8+1+3da3a3+db287c2a
To: "Chris McGarty" <sip:6787721321@10.212.7.226>;tag=as55882f8a
Call-ID: 1bcb90662f4a4e2379076fa35882452f@10.212.7.226
CSeq: 778024721 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6787721321@10.212.7.226>
Content-Length: 0


<------------>
Audio is at 10.212.7.226 port 19128
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
localhost*CLI>
<--- Transmitting (no NAT) to 211.126.32.246:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.126.32.246:5060;branch=z9hG4bK9uedvc30eg4h7c0r43o0sbepner80.1;received=211.126.32.246
From: <sip:14044084026@metaswitch.phoneprovider.com>;tag=SDe3mh899-10.246.15.8+1+3da3a3+db287c2a
To: "Chris McGarty" <sip:6787721321@10.212.7.226>;tag=as55882f8a
Call-ID: 1bcb90662f4a4e2379076fa35882452f@10.212.7.226
CSeq: 778024721 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6787721321@10.212.7.226>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 2584 2588 IN IP4 10.212.7.226
s=session
c=IN IP4 10.212.7.226
t=0 0
m=audio 19128 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
localhost*CLI>
<--- SIP read from 211.126.32.246:5060 --->
INVITE sip:6787721321@10.212.7.226 SIP/2.0
Via: SIP/2.0/UDP 211.126.32.246:5060;branch=z9hG4bK9uedvc30eg4h7c0r43o0sbepneb90.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info
Max-Forwards: 69
Call-ID: 1bcb90662f4a4e2379076fa35882452f@10.212.7.226
From: <sip:14044084026@metaswitch.phoneprovider.com>;tag=SDe3mh899-10.246.15.8+1+3da3a3+db287c2a
To: "Chris McGarty" <sip:6787721321@10.212.7.226>;tag=as55882f8a
CSeq: 778024722 INVITE
Expires: 180I>
Contact: <sip:14044084026@211.126.32.246:5060;transport=udp>
Supported: 100rel
Content-Length: 160
Content-Type: application/sdp

v=0
o=- 1554245977 1554245978 IN IP4 211.126.32.246
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 14842 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
--- (13 headers 8 lines) ---
localhost*CLI>
<--- Transmitting (no NAT) to 211.126.32.246:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 211.126.32.246:5060;branch=z9hG4bK9uedvc30eg4h7c0r43o0sbepneb90.1;received=211.126.32.246
From: <sip:14044084026@metaswitch.phoneprovider.com>;tag=SDe3mh899-10.246.15.8+1+3da3a3+db287c2a
To: "Chris McGarty" <sip:6787721321@10.212.7.226>;tag=as55882f8a
Call-ID: 1bcb90662f4a4e2379076fa35882452f@10.212.7.226
CSeq: 778024722 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


<------------>
localhost*CLI>
<--- SIP read from 211.126.32.246:5060 --->
ACK sip:6787721321@10.212.7.226 SIP/2.0
Via: SIP/2.0/UDP 211.126.32.246:5060;branch=z9hG4bK9uedvc30eg4h7c0r43o0sbepneb90.1
CSeq: 778024722 ACK
Max-Forwards: 69
Call-ID: 1bcb90662f4a4e2379076fa35882452f@10.212.7.226
From: <sip:14044084026@metaswitch.phoneprovider.com>;tag=SDe3mh899-10.246.15.8+1+3da3a3+db287c2a
To: "Chris McGarty" <sip:6787721321@10.212.7.226>;tag=as55882f8a
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
localhost*CLI>
<--- SIP read from 211.126.32.246:5060 --->
BYE sip:6787721321@10.212.7.226 SIP/2.0
Via: SIP/2.0/UDP 211.126.32.246:5060;branch=z9hG4bK9uedvc30eg4h7c0r43o0sdepner90.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info
Max-Forwards: 69
Call-ID: 1bcb90662f4a4e2379076fa35882452f@10.212.7.226
From: <sip:14044084026@metaswitch.phoneprovider.com>;tag=SDe3mh899-10.246.15.8+1+3da3a3+db287c2a
To: "Chris McGarty" <sip:6787721321@10.212.7.226>;tag=as55882f8a
CSeq: 778024723 BYE
Supported: 100rel
Content-Length: 0