<--- Transmitting (NAT) to 109.170.0.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 109.170.0.2:5060;branch=z9hG4bK1dde7294;received=109.170.0.2;rport=5060
From: "4957398818" <sip:4957398818@109.170.0.2>;tag=as1f471267
To: <sip:4997593014@10.33.105.138>;tag=as2ca24d0b
Call-ID: 719b7e0b088db72f0b87d0b90a46a8fe@109.170.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:4997593014@10.33.105.138>
Content-Length: 0


<------------>
    -- SIP/mostkom-0000bddd is making progress passing it to SIP/siptrunk-office-0000bddc
[Apr 28 20:18:16] NOTICE[29461]: rtp.c:831 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.20.127.65
    -- SIP/4997593014-0000bddf answered SIP/provru-0000bdde
Audio is at 10.33.105.138 port 13616
Adding codec 0x4 (ulaw) to SDP
gw1*CLI>
<--- Reliably Transmitting (NAT) to 109.170.0.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.170.0.2:5060;branch=z9hG4bK1dde7294;received=109.170.0.2;rport=5060
From: "4957398818" <sip:4957398818@109.170.0.2>;tag=as1f471267
To: <sip:4997593014@10.33.105.138>;tag=as2ca24d0b
Call-ID: 719b7e0b088db72f0b87d0b90a46a8fe@109.170.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:4997593014@10.33.105.138>
Content-Type: application/sdp
Content-Length: 186

v=0
o=root 18364 18364 IN IP4 10.33.105.138
s=session
c=IN IP4 10.33.105.138
t=0 0
m=audio 13616 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
gw1*CLI>
<--- SIP read from 109.170.0.2:5060 --->
ACK sip:4997593014@10.33.105.138 SIP/2.0
Via: SIP/2.0/UDP 109.170.0.2:5060;branch=z9hG4bK3033bf66;rport
Max-Forwards: 70
From: "4957398818" <sip:4957398818@109.170.0.2>;tag=as1f471267
To: <sip:4997593014@10.33.105.138>;tag=as2ca24d0b
Contact: <sip:4957398818@109.170.0.2>
Call-ID: 719b7e0b088db72f0b87d0b90a46a8fe@109.170.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- SIP/mostkom-0000bddd answered SIP/siptrunk-office-0000bddc
  == Spawn extension (incoming_calls, 4997593014, 5) exited non-zero on 'SIP/provru-0000bdde'
Scheduling destruction of SIP dialog '719b7e0b088db72f0b87d0b90a46a8fe@109.170.0.2' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:4957398818@109.170.0.2> for address/port to send to
set_destination: set destination to 109.170.0.2, port 5060
Reliably Transmitting (NAT) to 109.170.0.2:5060:
BYE sip:4957398818@109.170.0.2 SIP/2.0
Via: SIP/2.0/UDP 10.33.105.138:5060;branch=z9hG4bK17d6ae3e;rport
From: <sip:4997593014@10.33.105.138>;tag=as2ca24d0b
To: "4957398818" <sip:4957398818@109.170.0.2>;tag=as1f471267
Call-ID: 719b7e0b088db72f0b87d0b90a46a8fe@109.170.0.2
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
gw1*CLI>
<--- SIP read from 109.170.0.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.33.105.138:5060;branch=z9hG4bK17d6ae3e;received=10.33.105.138;rport=5060
From: <sip:4997593014@10.33.105.138>;tag=as2ca24d0b
To: "4957398818" <sip:4957398818@109.170.0.2>;tag=as1f471267
Call-ID: 719b7e0b088db72f0b87d0b90a46a8fe@109.170.0.2
CSeq: 102 BYE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


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