Untitled


<--- SIP read from UDP:64.2.142.214:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.27.170.251:5060;branch=z9hG4bK397f50c5;received=52.27.170.251
From: <sip:nwag_ctpbx@52.27.170.251>;tag=as4123fa29
To: <sip:5094349384@outbound.vitelity.net>;tag=as6d8c4c1c
Call-ID: 065cfd4f42fee041604fa97b40ab29fd@52.27.170.251:5060
CSeq: 103 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 32766 32767 IN IP4 64.2.142.214
s=session
c=IN IP4 64.2.142.214
t=0 0
m=audio 14310 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 16 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
failed to extend from 64 to 98
Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.2.142.214:14310
sip_route_dump: no route/path
[Nov 12 21:58:48] WARNING[4044][C-00000396]: chan_sip.c:16150 __set_address_from_contact: Invalid contact uri  (missing sip: or sips:), attempting to use anyway
[Nov 12 21:58:48] WARNING[4044][C-00000396]: chan_sip.c:16163 __set_address_from_contact: Invalid URI: parse_uri failed to acquire hostport
Transmitting (no NAT) to 64.2.142.214:5060:
ACK sip:5094349384@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 52.27.170.251:5060;branch=z9hG4bK371aaee9
Max-Forwards: 70
From: <sip:nwag_ctpbx@52.27.170.251>;tag=as4123fa29
To: <sip:5094349384@outbound.vitelity.net>;tag=as6d8c4c1c
Contact: <sip:nwag_ctpbx@52.27.170.251:5060>
Call-ID: 065cfd4f42fee041604fa97b40ab29fd@52.27.170.251:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
Reliably Transmitting (no NAT) to 64.2.142.214:5060:
BYE sip:5094349384@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 52.27.170.251:5060;branch=z9hG4bK35d4f251
Max-Forwards: 70
From: <sip:nwag_ctpbx@52.27.170.251>;tag=as4123fa29
To: <sip:5094349384@outbound.vitelity.net>;tag=as6d8c4c1c
Call-ID: 065cfd4f42fee041604fa97b40ab29fd@52.27.170.251:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 13.6.0
Proxy-Authorization: Digest username="nwag_ctpbx", realm="asterisk", algorithm=MD5, uri="sip:5094349384@outbound.vitelity.net", nonce="777ae00e", response="63e0937052dbcdbe8fec249fd01bb4fe"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---
Scheduling destruction of SIP dialog '065cfd4f42fee041604fa97b40ab29fd@52.27.170.251:5060' in 6400 ms (Method: INVITE)
    -- SIP/vitel-outbound-00001924 answered SIP/456-00001923
Audio is at 19982
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 173.160.189.58:55222 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 173.160.189.58:55222;branch=z9hG4bK-524287-1---cac4739bfe369f30;received=173.160.189.58;rport=55222
From: <sip:456@pbx.corporatetools.com;transport=UDP>;tag=f537db09
To: <sip:5555@pbx.corporatetools.com;transport=UDP>;tag=as314e0953
Call-ID: B43mUD3eTjobGtRiBC2eeg..
CSeq: 2 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5555@52.27.170.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 288

v=0
o=root 1171928443 1171928443 IN IP4 52.27.170.251
s=Asterisk PBX 13.6.0
c=IN IP4 52.27.170.251
t=0 0
m=audio 19982 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/vitel-outbound-00001924 joined 'simple_bridge' basic-bridge <9e3716c4-b375-4cf6-be8f-2acada327271>
    -- Channel SIP/456-00001923 joined 'simple_bridge' basic-bridge <9e3716c4-b375-4cf6-be8f-2acada327271>
       > Bridge 9e3716c4-b375-4cf6-be8f-2acada327271: switching from simple_bridge technology to native_rtp
       > Locally RTP bridged 'SIP/456-00001923' and 'SIP/vitel-outbound-00001924' in stack
       > Locally RTP bridged 'SIP/456-00001923' and 'SIP/vitel-outbound-00001924' in stack
    -- Channel SIP/vitel-outbound-00001924 left 'native_rtp' basic-bridge <9e3716c4-b375-4cf6-be8f-2acada327271>
    -- Channel SIP/456-00001923 left 'native_rtp' basic-bridge <9e3716c4-b375-4cf6-be8f-2acada327271>
  == Spawn extension (from-internal, 5555, 1) exited non-zero on 'SIP/456-00001923'
Scheduling destruction of SIP dialog 'B43mUD3eTjobGtRiBC2eeg..' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '065cfd4f42fee041604fa97b40ab29fd@52.27.170.251:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 64.2.142.214:5060:
BYE sip:5094349384@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 52.27.170.251:5060;branch=z9hG4bK518c6469
Max-Forwards: 70
From: <sip:nwag_ctpbx@52.27.170.251>;tag=as4123fa29
To: <sip:5094349384@outbound.vitelity.net>;tag=as6d8c4c1c
Call-ID: 065cfd4f42fee041604fa97b40ab29fd@52.27.170.251:5060
CSeq: 105 BYE
User-Agent: Asterisk PBX 13.6.0
Proxy-Authorization: Digest username="nwag_ctpbx", realm="asterisk", algorithm=MD5, uri="sip:5094349384@outbound.vitelity.net", nonce="777ae00e", response="63e0937052dbcdbe8fec249fd01bb4fe"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
ip-172-31-26-192*CLI>

<--- SIP read from UDP:64.2.142.214:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.27.170.251:5060;branch=z9hG4bK35d4f251;received=52.27.170.251
From: <sip:nwag_ctpbx@52.27.170.251>;tag=as4123fa29
To: <sip:5094349384@outbound.vitelity.net>;tag=as6d8c4c1c
Call-ID: 065cfd4f42fee041604fa97b40ab29fd@52.27.170.251:5060
CSeq: 104 BYE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:173.160.189.58:55222 --->
ACK sip:5555@52.27.170.251:5060 SIP/2.0
Via: SIP/2.0/UDP 173.160.189.58:55222;branch=z9hG4bK-524287-1---81489759d51d1bd8;rport
Max-Forwards: 70
Contact: <sip:456@173.160.189.58:55222;transport=UDP>
To: <sip:5555@pbx.corporatetools.com;transport=UDP>;tag=as314e0953
From: <sip:456@pbx.corporatetools.com;transport=UDP>;tag=f537db09
Call-ID: B43mUD3eTjobGtRiBC2eeg..
CSeq: 2 ACK
User-Agent: Zoiper r33688
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Reliably Transmitting (NAT) to 173.160.189.58:55222:
BYE sip:456@173.160.189.58:55222;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 52.27.170.251:5060;branch=z9hG4bK1c65bb45;rport
Max-Forwards: 70
From: <sip:5555@pbx.corporatetools.com;transport=UDP>;tag=as314e0953
To: <sip:456@pbx.corporatetools.com;transport=UDP>;tag=f537db09
Call-ID: B43mUD3eTjobGtRiBC2eeg..
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.6.0
Proxy-Authorization: Digest username="456", realm="asterisk", algorithm=MD5, uri="sip:pbx.corporatetools.com", nonce="3d897962", response="1be24bf8c9aaf8b413d88fcad4ed0262"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog 'B43mUD3eTjobGtRiBC2eeg..' in 6400 ms (Method: ACK)

<--- SIP read from UDP:64.2.142.214:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 52.27.170.251:5060;branch=z9hG4bK518c6469;received=52.27.170.251
From: <sip:nwag_ctpbx@52.27.170.251>;tag=as4123fa29
To: <sip:5094349384@outbound.vitelity.net>;tag=as6d8c4c1c
Call-ID: 065cfd4f42fee041604fa97b40ab29fd@52.27.170.251:5060
CSeq: 105 BYE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '065cfd4f42fee041604fa97b40ab29fd@52.27.170.251:5060' Method: INVITE
ip-172-31-26-192*CLI>

<--- SIP read from UDP:173.160.189.58:55222 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.27.170.251:5060;branch=z9hG4bK1c65bb45;rport=5060
Contact: <sip:456@173.160.189.58:55222;transport=UDP>
To: <sip:456@pbx.corporatetools.com;transport=UDP>;tag=f537db09
From: <sip:5555@pbx.corporatetools.com;transport=UDP>;tag=as314e0953
Call-ID: B43mUD3eTjobGtRiBC2eeg..
CSeq: 102 BYE
User-Agent: Zoiper r33688
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'B43mUD3eTjobGtRiBC2eeg..' Method: ACK