Connected to Asterisk 13.3.2 currently running on almudo (pid = 8286)
  == Using SIP RTP CoS mark 5
    -- Executing [5659@from-internal:1] NoOp("SIP/enlace-trunk-00000002", "Incomming call from 5659 to "2288175116" <2288175116> to PAranoids") in new stack
    -- Executing [5659@from-internal:2] Set("SIP/enlace-trunk-00000002", "CALLERID(all)=9999") in new stack
    -- Executing [5659@from-internal:3] Dial("SIP/enlace-trunk-00000002", "SIP/paralax/5659") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/paralax/5659
    -- SIP/paralax-00000003 is ringing
    -- SIP/paralax-00000003 answered SIP/enlace-trunk-00000002
    -- Channel SIP/enlace-trunk-00000002 joined 'simple_bridge' basic-bridge <1e6f92e5-b154-43e9-a729-9cc184b6c4f2>
    -- Channel SIP/paralax-00000003 joined 'simple_bridge' basic-bridge <1e6f92e5-b154-43e9-a729-9cc184b6c4f2>
       > Bridge 1e6f92e5-b154-43e9-a729-9cc184b6c4f2: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'SIP/paralax-00000003' and 'SIP/enlace-trunk-00000002' - media will flow directly between them
       > Remotely bridged 'SIP/paralax-00000003' and 'SIP/enlace-trunk-00000002' - media will flow directly between them
    -- Channel SIP/paralax-00000003 left 'native_rtp' basic-bridge <1e6f92e5-b154-43e9-a729-9cc184b6c4f2>
    -- Channel SIP/enlace-trunk-00000002 left 'native_rtp' basic-bridge <1e6f92e5-b154-43e9-a729-9cc184b6c4f2>
  == Spawn extension (from-internal, 5659, 3) exited non-zero on 'SIP/enlace-trunk-00000002'
almudo*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:10.100.5.67:5060 --->
INVITE sip:5659@sbcacme.ims.iusatel.com:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKibh86f200o9hv4lli6a0.1
To: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>
From: "2288175116"<sip:2288175116@10.208.19.193;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 1 INVITE
Max-Forwards: 66
Content-Length: 333
Contact: <sip:10.100.5.67:5060;transport=udp>
Content-Type: application/sdp
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, INFO, PRACK, NOTIFY, MESSAGE, UPDATE
Supported: 100rel, timer, histinfo, precondition
P-Asserted-Identity: <sip:2288175116@SGDA100NNGSMMBA.iusatel.com;SBC;SBC>
Min-SE: 90
Session-Expires: 1800;refresher=uac
P-Early-Media: supported
Route: <sip:2282375659@10.50.98.214:5060;user=phone;lr>

v=0
o=HuaweiSoftx3000 1109197089 1109197090 IN IP4 10.100.5.67
s=SipCall
c=IN IP4 10.100.5.67
t=0 0
m=audio 59534 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=ptime:20
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
--- (17 headers 14 lines) ---
Sending to 10.100.5.67:5060 (no NAT)
Sending to 10.100.5.67:5060 (no NAT)
Using INVITE request as basis request - yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
Found peer 'enlace-trunk' for '2288175116' from 10.100.5.67:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 116
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 116
Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.100.5.67:59534
Looking for 5659 in from-internal (domain sbcacme.ims.iusatel.com)
sip_route_dump: route/path hop: <sip:10.100.5.67:5060;transport=udp>

<--- Transmitting (no NAT) to 10.100.5.67:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKibh86f200o9hv4lli6a0.1;received=10.100.5.67
From: "2288175116"<sip:2288175116@10.208.19.193;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
To: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:5659@10.50.98.214:5060>
Content-Length: 0


<------------>
    -- Executing [5659@from-internal:1] NoOp("SIP/enlace-trunk-00000004", "Incomming call from 5659 to "2288175116" <2288175116> to PAranoids") in new stack
    -- Executing [5659@from-internal:2] Set("SIP/enlace-trunk-00000004", "CALLERID(all)=9999") in new stack
    -- Executing [5659@from-internal:3] Dial("SIP/enlace-trunk-00000004", "SIP/paralax/5659") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11086
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.1.251:5060:
INVITE sip:5659@10.0.1.251 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7bec7829
Max-Forwards: 70
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>
Contact: <sip:9999@10.0.1.115:5060>
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.3.2
Date: Mon, 08 Feb 2016 17:06:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1718921788 1718921788 IN IP4 10.0.1.115
s=Asterisk PBX 13.3.2
c=IN IP4 10.0.1.115
t=0 0
m=audio 11086 RTP/AVP 8 0 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/paralax/5659

<--- SIP read from UDP:10.0.1.251:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7bec7829;received=10.0.1.115;rport=5060
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>;tag=as0b69474c
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 102 INVITE
Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="187.188.213.132", nonce="24d8e0eb"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 10.0.1.251:5060:
ACK sip:5659@10.0.1.251 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7bec7829
Max-Forwards: 70
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>;tag=as0b69474c
Contact: <sip:9999@10.0.1.115:5060>
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.3.2
Content-Length: 0


---
Audio is at 11086
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.1.251:5060:
INVITE sip:5659@10.0.1.251 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK57e27115
Max-Forwards: 70
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>
Contact: <sip:9999@10.0.1.115:5060>
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.3.2
Authorization: Digest username="9999", realm="187.188.213.132", algorithm=MD5, uri="sip:5659@10.0.1.251", nonce="24d8e0eb", response="ad75953f2fb5e5512a246c493ef10fc3"
Date: Mon, 08 Feb 2016 17:06:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1718921788 1718921789 IN IP4 10.0.1.115
s=Asterisk PBX 13.3.2
c=IN IP4 10.0.1.115
t=0 0
m=audio 11086 RTP/AVP 8 0 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:10.0.1.251:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK57e27115;received=10.0.1.115;rport=5060
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 103 INVITE
Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5659@10.0.1.251:5060>
Content-Length: 0

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

<--- SIP read from UDP:10.0.1.251:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK57e27115;received=10.0.1.115;rport=5060
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>;tag=as4f65d3d3
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 103 INVITE
Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5659@10.0.1.251:5060>
P-Asserted-Identity: "dev8" <sip:9998@10.0.1.115>
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:5659@10.0.1.251:5060>
    -- SIP/paralax-00000005 is ringing

<--- Transmitting (no NAT) to 10.100.5.67:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKibh86f200o9hv4lli6a0.1;received=10.100.5.67
From: "2288175116"<sip:2288175116@10.208.19.193;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
To: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:5659@10.50.98.214:5060>
Content-Length: 0


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

<--- SIP read from UDP:10.0.1.251:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK57e27115;received=10.0.1.115;rport=5060
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>;tag=as4f65d3d3
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 103 INVITE
Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5659@10.0.1.251:5060>
P-Asserted-Identity: "dev8" <sip:9998@10.0.1.115>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 1402105843 1402105843 IN IP4 10.0.1.251
s=Asterisk PBX 13.0.1
c=IN IP4 10.0.1.251
t=0 0
m=audio 15450 RTP/AVP 8 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=maxptime:150
a=sendrecv
<------------->
--- (15 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 116
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 116
Capabilities: us - (alaw|ulaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.1.251:15450
sip_route_dump: route/path hop: <sip:5659@10.0.1.251:5060>
set_destination: Parsing <sip:5659@10.0.1.251:5060> for address/port to send to
set_destination: set destination to 10.0.1.251:5060
Transmitting (no NAT) to 10.0.1.251:5060:
ACK sip:5659@10.0.1.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK3f3b65a1
Max-Forwards: 70
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>;tag=as4f65d3d3
Contact: <sip:9999@10.0.1.115:5060>
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.3.2
Content-Length: 0


---
    -- SIP/paralax-00000005 answered SIP/enlace-trunk-00000004
Audio is at 13642
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.100.5.67:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKibh86f200o9hv4lli6a0.1;received=10.100.5.67
From: "2288175116"<sip:2288175116@10.208.19.193;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
To: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 1 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:5659@10.50.98.214:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 284

v=0
o=root 438711394 438711394 IN IP4 10.50.98.214
s=Asterisk PBX 13.3.2
c=IN IP4 10.50.98.214
t=0 0
m=audio 13642 RTP/AVP 8 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/enlace-trunk-00000004 joined 'simple_bridge' basic-bridge <041d4c74-5a1e-44d3-99da-77bffd26259b>
    -- Channel SIP/paralax-00000005 joined 'simple_bridge' basic-bridge <041d4c74-5a1e-44d3-99da-77bffd26259b>
       > Bridge 041d4c74-5a1e-44d3-99da-77bffd26259b: switching from simple_bridge technology to native_rtp
set_destination: Parsing <sip:5659@10.0.1.251:5060> for address/port to send to
set_destination: set destination to 10.0.1.251:5060
Audio is at 11086
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.1.251:5060:
INVITE sip:5659@10.0.1.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7160ff0f
Max-Forwards: 70
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>;tag=as4f65d3d3
Contact: <sip:9999@10.0.1.115:5060>
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 13.3.2
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1718921788 1718921790 IN IP4 10.100.5.67
s=Asterisk PBX 13.3.2
c=IN IP4 10.100.5.67
t=0 0
m=audio 59534 RTP/AVP 8 0 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=maxptime:150
a=sendrecv

---
       > Remotely bridged 'SIP/paralax-00000005' and 'SIP/enlace-trunk-00000004' - media will flow directly between them
       > Remotely bridged 'SIP/paralax-00000005' and 'SIP/enlace-trunk-00000004' - media will flow directly between them

<--- SIP read from UDP:10.0.1.251:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7160ff0f;received=10.0.1.115;rport=5060
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>;tag=as4f65d3d3
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 104 INVITE
Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5659@10.0.1.251:5060>
Content-Length: 0

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

<--- SIP read from UDP:10.0.1.251:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7160ff0f;received=10.0.1.115;rport=5060
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>;tag=as4f65d3d3
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 104 INVITE
Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5659@10.0.1.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 282

v=0
o=root 1402105843 1402105844 IN IP4 10.0.1.251
s=Asterisk PBX 13.0.1
c=IN IP4 10.0.1.251
t=0 0
m=audio 15450 RTP/AVP 8 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=maxptime:150
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 116
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 116
Capabilities: us - (alaw|ulaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.1.251:15450
set_destination: Parsing <sip:5659@10.0.1.251:5060> for address/port to send to
set_destination: set destination to 10.0.1.251:5060
Transmitting (no NAT) to 10.0.1.251:5060:
ACK sip:5659@10.0.1.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7ffb1e69
Max-Forwards: 70
From: <sip:9999@10.0.1.115>;tag=as5e212125
To: <sip:5659@10.0.1.251>;tag=as4f65d3d3
Contact: <sip:9999@10.0.1.115:5060>
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 13.3.2
Content-Length: 0


---

<--- SIP read from UDP:10.100.5.67:5060 --->
ACK sip:5659@10.50.98.214:5060 SIP/2.0
Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKdstbld3008ugh3hvn100.1
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
From: "2288175116"<sip:2288175116@empresarialtk.iusacell.com;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
To: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
CSeq: 1 ACK
Max-Forwards: 68
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.100.5.67:5060
Audio is at 13642
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.100.5.67:5060:
INVITE sip:10.100.5.67:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK3557de89
Max-Forwards: 70
From: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
To: "2288175116"<sip:2288175116@10.208.19.193;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
Contact: <sip:5659@10.50.98.214:5060>
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.3.2
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 438711394 438711395 IN IP4 10.0.1.251
s=Asterisk PBX 13.3.2
c=IN IP4 10.0.1.251
t=0 0
m=audio 15450 RTP/AVP 8 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:10.100.5.67:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK3557de89
From: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
To: "2288175116"<sip:2288175116@10.208.19.193;transport=udp>;tag=go5z55gn-CC-1014-OFC-231
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 102 INVITE
Require: timer
Session-Expires: 1800;refresher=uas
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Contact: <sip:10.100.5.67:5060;transport=udp>
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1109197089 1109197091 IN IP4 10.100.5.67
s=SipCall
c=IN IP4 10.100.5.67
t=0 0
m=audio 59534 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=ptime:20
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 116
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 116
Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.100.5.67:59534
set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.100.5.67:5060
Transmitting (no NAT) to 10.100.5.67:5060:
ACK sip:10.100.5.67:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK56958df8
Max-Forwards: 70
From: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
To: "2288175116"<sip:2288175116@10.208.19.193;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
Contact: <sip:5659@10.50.98.214:5060>
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.3.2
Content-Length: 0


---

<--- SIP read from UDP:10.0.1.251:5060 --->
BYE sip:9999@10.0.1.115:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.251:5060;branch=z9hG4bK2cf673ec;rport
Max-Forwards: 70
From: <sip:5659@10.0.1.251>;tag=as4f65d3d3
To: <sip:9999@10.0.1.115>;tag=as5e212125
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 102 BYE
User-Agent: FPBX-AsteriskNOW-12.0.68(13.0.1)
Proxy-Authorization: Digest username="9999", realm="187.188.213.132", algorithm=MD5, uri="sip:10.0.1.251", nonce="24d8e0eb", response="7e7221853f1e2f04b7023a0b56faa86c"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 10.0.1.251:5060 (no NAT)
Scheduling destruction of SIP dialog '4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.0.1.251:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.251:5060;branch=z9hG4bK2cf673ec;received=10.0.1.251;rport=5060
From: <sip:5659@10.0.1.251>;tag=as4f65d3d3
To: <sip:9999@10.0.1.115>;tag=as5e212125
Call-ID: 4c9665ca04b104ff0e4d77f01e16a3fa@10.0.1.115:5060
CSeq: 102 BYE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/paralax-00000005 left 'native_rtp' basic-bridge <041d4c74-5a1e-44d3-99da-77bffd26259b>
set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.100.5.67:5060
Audio is at 13642
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.100.5.67:5060:
INVITE sip:10.100.5.67:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK25efe9e2
Max-Forwards: 70
From: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
To: "2288175116"<sip:2288175116@10.208.19.193;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
Contact: <sip:5659@10.50.98.214:5060>
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.3.2
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 438711394 438711396 IN IP4 10.50.98.214
s=Asterisk PBX 13.3.2
c=IN IP4 10.50.98.214
t=0 0
m=audio 13642 RTP/AVP 8 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=maxptime:150
a=sendrecv

---
    -- Channel SIP/enlace-trunk-00000004 left 'native_rtp' basic-bridge <041d4c74-5a1e-44d3-99da-77bffd26259b>
  == Spawn extension (from-internal, 5659, 3) exited non-zero on 'SIP/enlace-trunk-00000004'
Scheduling destruction of SIP dialog 'yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64' in 32000 ms (Method: ACK)

<--- SIP read from UDP:10.100.5.67:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK25efe9e2
From: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
To: "2288175116"<sip:2288175116@10.208.19.193;transport=udp>;tag=go5z55gn-CC-1014-OFC-231
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 103 INVITE
Require: timer
Session-Expires: 1800;refresher=uas
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Contact: <sip:10.100.5.67:5060;transport=udp>
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1109197089 1109197092 IN IP4 10.100.5.67
s=SipCall
c=IN IP4 10.100.5.67
t=0 0
m=audio 59534 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=ptime:20
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 116
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 116
Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.100.5.67:59534
set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.100.5.67:5060
Transmitting (no NAT) to 10.100.5.67:5060:
ACK sip:10.100.5.67:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK60e7eadd
Max-Forwards: 70
From: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
To: "2288175116"<sip:2288175116@10.208.19.193;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
Contact: <sip:5659@10.50.98.214:5060>
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.3.2
Content-Length: 0


---
set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.100.5.67:5060
Reliably Transmitting (no NAT) to 10.100.5.67:5060:
BYE sip:10.100.5.67:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK473307f5
Max-Forwards: 70
From: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
To: "2288175116"<sip:2288175116@10.208.19.193;transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 104 BYE
User-Agent: Asterisk PBX 13.3.2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog 'yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64' in 32000 ms (Method: ACK)

<--- SIP read from UDP:10.100.5.67:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK473307f5
From: "1331342282375659"<sip:2375659@10.0.104.66;transport=udp>;tag=as7723fa42
To: "2288175116"<sip:2288175116@10.208.19.193;transport=udp>;tag=go5z55gn-CC-1014-OFC-231
Call-ID: yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64
CSeq: 104 BYE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'yuzz67g6s96gu3n4s9s4sapnst9ssu9t@10.18.5.64' Method: ACK