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THUFIR/failed_call_siptrunk_DID

Created July 4, 2016 04:24
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failed_call_to_siptrunk_DID
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failed_call_siptrunk_DID
mordor*CLI>
mordor*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:192.168.1.5:5062 --->
NOTIFY sip:192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-3dfaf8af
From: "thufir" <sip:thufir@192.168.1.8>;tag=2dc2ec8b9eeabd55o2
To: <sip:192.168.1.8>
Call-ID: 8361cdb3-79410e4d@192.168.1.5
CSeq: 59 NOTIFY
Max-Forwards: 70
Contact: "thufir" <sip:thufir@192.168.1.5:5062>
Event: keep-alive
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- Transmitting (NAT) to 192.168.1.5:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-3dfaf8af;received=192.168.1.5;rport=5062
From: "thufir" <sip:thufir@192.168.1.8>;tag=2dc2ec8b9eeabd55o2
To: <sip:192.168.1.8>;tag=as35108297
Call-ID: 8361cdb3-79410e4d@192.168.1.5
CSeq: 59 NOTIFY
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '8361cdb3-79410e4d@192.168.1.5' in 32000 ms (Method: NOTIFY)
<--- SIP read from UDP:192.168.1.5:5063 --->
NOTIFY sip:192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-9b54d938
From: "piter" <sip:piter@192.168.1.8>;tag=426902db72b1b45o3
To: <sip:192.168.1.8>
Call-ID: 8024367b-67307c25@192.168.1.5
CSeq: 59 NOTIFY
Max-Forwards: 70
Contact: "piter" <sip:piter@192.168.1.5:5063>
Event: keep-alive
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- Transmitting (NAT) to 192.168.1.5:5063 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-9b54d938;received=192.168.1.5;rport=5063
From: "piter" <sip:piter@192.168.1.8>;tag=426902db72b1b45o3
To: <sip:192.168.1.8>;tag=as7f962db4
Call-ID: 8024367b-67307c25@192.168.1.5
CSeq: 59 NOTIFY
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '8024367b-67307c25@192.168.1.5' in 32000 ms (Method: NOTIFY)
<--- SIP read from UDP:63.247.69.226:5060 --->
INVITE sip:12345678901@123.456.789.123:5060 SIP/2.0
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Max-Forwards: 32
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Accept: application/sdp
Contact: "earthling" <sip:+19876543210@67.231.5.110:5060>
Supported: replaces
Content-Length: 281
Content-Disposition: session; handling=required
Content-Type: application/sdp
Privacy: off
v=0
o=Sonus_UAC 786815 776798 IN IP4 67.231.5.110
s=SIP Media Capabilities
c=IN IP4 67.231.5.76
t=0 0
m=audio 33718 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
--- (17 headers 13 lines) ---
Sending to 63.247.69.226:5060 (NAT)
Sending to 63.247.69.226:5060 (NAT)
Using INVITE request as basis request - 1342742344_134122685@67.231.5.110
Found peer '65412378GW1' for '9876543210' from 63.247.69.226:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 67.231.5.76:33718
Looking for 12345678901 in inbound (domain 123.456.789.123)
sip_route_dump: route/path hop: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
<--- Transmitting (NAT) to 63.247.69.226:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0;received=63.247.69.226;rport=5060
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12345678901@123.456.789.123:5060>
Content-Length: 0
<------------>
-- Executing [12345678901@inbound:1] NoOp("SIP/65412378GW1-00000002", "") in new stack
-- Executing [12345678901@inbound:2] Dial("SIP/65412378GW1-00000002", "SIP/thufir,60") in new stack
== Using SIP RTP CoS mark 5
Audio is at 15238
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 192.168.1.5:5062:
INVITE sip:thufir@192.168.1.5:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK5b5ce5b5;rport
Max-Forwards: 70
From: "PLANET EARTH " <sip:9876543210@192.168.1.8>;tag=as5042a3ac
To: <sip:thufir@192.168.1.5:5062>
Contact: <sip:9876543210@192.168.1.8:5060>
Call-ID: 4f4a94fa0f17d1551440124405b35018@192.168.1.8:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Date: Mon, 04 Jul 2016 04:05:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 2032774532 2032774532 IN IP4 192.168.1.8
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 192.168.1.8
t=0 0
m=audio 15238 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=ptime:20
a=maxptime:150
a=sendrecv
---
-- Called SIP/thufir
<--- SIP read from UDP:192.168.1.5:5062 --->
SIP/2.0 100 Trying
To: <sip:thufir@192.168.1.5:5062>
From: "PLANET EARTH " <sip:9876543210@192.168.1.8>;tag=as5042a3ac
Call-ID: 4f4a94fa0f17d1551440124405b35018@192.168.1.8:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK5b5ce5b5
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.5:5062 --->
SIP/2.0 180 Ringing
To: <sip:thufir@192.168.1.5:5062>;tag=441ea2b92b29287ei2
From: "PLANET EARTH " <sip:9876543210@192.168.1.8>;tag=as5042a3ac
Call-ID: 4f4a94fa0f17d1551440124405b35018@192.168.1.8:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK5b5ce5b5
Contact: "thufir" <sip:thufir@192.168.1.5:5062>
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:thufir@192.168.1.5:5062>
-- SIP/thufir-00000003 is ringing
<--- Transmitting (NAT) to 63.247.69.226:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0;received=63.247.69.226;rport=5060
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12345678901@123.456.789.123:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.5:5062 --->
SIP/2.0 200 OK
To: <sip:thufir@192.168.1.5:5062>;tag=441ea2b92b29287ei2
From: "PLANET EARTH " <sip:9876543210@192.168.1.8>;tag=as5042a3ac
Call-ID: 4f4a94fa0f17d1551440124405b35018@192.168.1.8:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK5b5ce5b5
Contact: "thufir" <sip:thufir@192.168.1.5:5062>
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 202
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 90468 90468 IN IP4 192.168.1.5
s=-
c=IN IP4 192.168.1.5
t=0 0
m=audio 16478 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.5:16478
sip_route_dump: route/path hop: <sip:thufir@192.168.1.5:5062>
Transmitting (NAT) to 192.168.1.5:5062:
ACK sip:thufir@192.168.1.5:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK318707ef;rport
Max-Forwards: 70
From: "PLANET EARTH " <sip:9876543210@192.168.1.8>;tag=as5042a3ac
To: <sip:thufir@192.168.1.5:5062>;tag=441ea2b92b29287ei2
Contact: <sip:9876543210@192.168.1.8:5060>
Call-ID: 4f4a94fa0f17d1551440124405b35018@192.168.1.8:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Content-Length: 0
---
-- SIP/thufir-00000003 answered SIP/65412378GW1-00000002
Audio is at 17772
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 63.247.69.226:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0;received=63.247.69.226;rport=5060
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12345678901@123.456.789.123:5060>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 323107781 323107781 IN IP4 123.456.789.123
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 123.456.789.123
t=0 0
m=audio 17772 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=ptime:20
a=maxptime:150
a=sendrecv
<------------>
-- Channel SIP/65412378GW1-00000002 joined 'simple_bridge' basic-bridge <0ee6a0de-c54a-4117-80a4-5d137d97234f>
-- Channel SIP/thufir-00000003 joined 'simple_bridge' basic-bridge <0ee6a0de-c54a-4117-80a4-5d137d97234f>
> Bridge 0ee6a0de-c54a-4117-80a4-5d137d97234f: switching from simple_bridge technology to native_rtp
> 0x7f0dc8005870 -- Probation passed - setting RTP source address to 192.168.1.5:16478
Retransmitting #1 (NAT) to 63.247.69.226:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0;received=63.247.69.226;rport=5060
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12345678901@123.456.789.123:5060>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 323107781 323107781 IN IP4 123.456.789.123
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 123.456.789.123
t=0 0
m=audio 17772 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=ptime:20
a=maxptime:150
a=sendrecv
---
> 0x7f0db000a2b0 -- Probation passed - setting RTP source address to 67.231.5.76:33718
Retransmitting #2 (NAT) to 63.247.69.226:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0;received=63.247.69.226;rport=5060
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12345678901@123.456.789.123:5060>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 323107781 323107781 IN IP4 123.456.789.123
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 123.456.789.123
t=0 0
m=audio 17772 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=ptime:20
a=maxptime:150
a=sendrecv
---
Retransmitting #3 (NAT) to 63.247.69.226:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0;received=63.247.69.226;rport=5060
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12345678901@123.456.789.123:5060>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 323107781 323107781 IN IP4 123.456.789.123
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 123.456.789.123
t=0 0
m=audio 17772 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=ptime:20
a=maxptime:150
a=sendrecv
---
Retransmitting #4 (NAT) to 63.247.69.226:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0;received=63.247.69.226;rport=5060
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12345678901@123.456.789.123:5060>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 323107781 323107781 IN IP4 123.456.789.123
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 123.456.789.123
t=0 0
m=audio 17772 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=ptime:20
a=maxptime:150
a=sendrecv
---
Retransmitting #5 (NAT) to 63.247.69.226:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0;received=63.247.69.226;rport=5060
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12345678901@123.456.789.123:5060>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 323107781 323107781 IN IP4 123.456.789.123
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 123.456.789.123
t=0 0
m=audio 17772 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=ptime:20
a=maxptime:150
a=sendrecv
---
Reliably Transmitting (NAT) to 192.168.1.5:5063:
OPTIONS sip:piter@192.168.1.5:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6d591156;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.8>;tag=as00caac37
To: <sip:piter@192.168.1.5:5063>
Contact: <sip:asterisk@192.168.1.8:5060>
Call-ID: 6ae67c8d7cef353f206002d34ee517ff@192.168.1.8:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Date: Mon, 04 Jul 2016 04:05:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
---
<--- SIP read from UDP:192.168.1.5:5063 --->
SIP/2.0 200 OK
To: <sip:piter@192.168.1.5:5063>;tag=58cb13139b39886di3
From: "asterisk" <sip:asterisk@192.168.1.8>;tag=as00caac37
Call-ID: 6ae67c8d7cef353f206002d34ee517ff@192.168.1.8:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6d591156
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '6ae67c8d7cef353f206002d34ee517ff@192.168.1.8:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.5:5062:
OPTIONS sip:thufir@192.168.1.5:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK4fdc92a7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.8>;tag=as5e9b6669
To: <sip:thufir@192.168.1.5:5062>
Contact: <sip:asterisk@192.168.1.8:5060>
Call-ID: 57c860be75d2ddb06a1e53f12f92e7b9@192.168.1.8:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Date: Mon, 04 Jul 2016 04:05:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
---
<--- SIP read from UDP:192.168.1.5:5062 --->
SIP/2.0 200 OK
To: <sip:thufir@192.168.1.5:5062>;tag=cfefcd434af893fdi2
From: "asterisk" <sip:asterisk@192.168.1.8>;tag=as5e9b6669
Call-ID: 57c860be75d2ddb06a1e53f12f92e7b9@192.168.1.8:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK4fdc92a7
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '57c860be75d2ddb06a1e53f12f92e7b9@192.168.1.8:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 63.247.69.226:5060:
OPTIONS sip:gw1.siptrunk.com SIP/2.0
Via: SIP/2.0/UDP 123.456.789.123:5060;branch=z9hG4bK482d0e6d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@123.456.789.123>;tag=as13c82adb
To: <sip:gw1.siptrunk.com>
Contact: <sip:asterisk@123.456.789.123:5060>
Call-ID: 2c3d2a8e0a7cb10b314e523b370b8d13@123.456.789.123:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Date: Mon, 04 Jul 2016 04:05:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
---
Reliably Transmitting (NAT) to 63.247.69.226:5060:
OPTIONS sip:gw1.siptrunk.com SIP/2.0
Via: SIP/2.0/UDP 123.456.789.123:5060;branch=z9hG4bK50f774d2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@123.456.789.123>;tag=as7ef4c0db
To: <sip:gw1.siptrunk.com>
Contact: <sip:asterisk@123.456.789.123:5060>
Call-ID: 637bc10a4c6d1f0446d232043a916b13@123.456.789.123:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Date: Mon, 04 Jul 2016 04:05:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
---
<--- SIP read from UDP:63.247.69.226:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.456.789.123:5060;branch=z9hG4bK482d0e6d;rport=49192;received=123.456.789.123
From: "asterisk" <sip:asterisk@123.456.789.123>;tag=as13c82adb
To: <sip:gw1.siptrunk.com>;tag=14575c43173f48ab35024af4364ef5fa.c2b7
Call-ID: 2c3d2a8e0a7cb10b314e523b370b8d13@123.456.789.123:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '2c3d2a8e0a7cb10b314e523b370b8d13@123.456.789.123:5060' Method: OPTIONS
<--- SIP read from UDP:63.247.69.226:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.456.789.123:5060;branch=z9hG4bK50f774d2;rport=49192;received=123.456.789.123
From: "asterisk" <sip:asterisk@123.456.789.123>;tag=as7ef4c0db
To: <sip:gw1.siptrunk.com>;tag=14575c43173f48ab35024af4364ef5fa.6f54
Call-ID: 637bc10a4c6d1f0446d232043a916b13@123.456.789.123:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '637bc10a4c6d1f0446d232043a916b13@123.456.789.123:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.1.5:5062 --->
NOTIFY sip:192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-b6766c8a
From: "thufir" <sip:thufir@192.168.1.8>;tag=2dc2ec8b9eeabd55o2
To: <sip:192.168.1.8>
Call-ID: 8361cdb3-79410e4d@192.168.1.5
CSeq: 60 NOTIFY
Max-Forwards: 70
Contact: "thufir" <sip:thufir@192.168.1.5:5062>
Event: keep-alive
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- Transmitting (NAT) to 192.168.1.5:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-b6766c8a;received=192.168.1.5;rport=5062
From: "thufir" <sip:thufir@192.168.1.8>;tag=2dc2ec8b9eeabd55o2
To: <sip:192.168.1.8>;tag=as35108297
Call-ID: 8361cdb3-79410e4d@192.168.1.5
CSeq: 60 NOTIFY
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '8361cdb3-79410e4d@192.168.1.5' in 32000 ms (Method: NOTIFY)
<--- SIP read from UDP:192.168.1.5:5063 --->
NOTIFY sip:192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-3e2304f5
From: "piter" <sip:piter@192.168.1.8>;tag=426902db72b1b45o3
To: <sip:192.168.1.8>
Call-ID: 8024367b-67307c25@192.168.1.5
CSeq: 60 NOTIFY
Max-Forwards: 70
Contact: "piter" <sip:piter@192.168.1.5:5063>
Event: keep-alive
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- Transmitting (NAT) to 192.168.1.5:5063 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-3e2304f5;received=192.168.1.5;rport=5063
From: "piter" <sip:piter@192.168.1.8>;tag=426902db72b1b45o3
To: <sip:192.168.1.8>;tag=as7f962db4
Call-ID: 8024367b-67307c25@192.168.1.5
CSeq: 60 NOTIFY
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '8024367b-67307c25@192.168.1.5' in 32000 ms (Method: NOTIFY)
Retransmitting #6 (NAT) to 63.247.69.226:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.247.69.226;branch=z9hG4bKbab8.6df23869e60577de3ee6cd6c58696600.0;received=63.247.69.226;rport=5060
Via: SIP/2.0/UDP 67.231.5.110:5060;branch=z9hG4bK08Bb8f4f3598cc01247
Record-Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
From: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
To: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 591368 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:12345678901@123.456.789.123:5060>
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 323107781 323107781 IN IP4 123.456.789.123
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 123.456.789.123
t=0 0
m=audio 17772 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=ptime:20
a=maxptime:150
a=sendrecv
---
[Jul 3 21:05:42] WARNING[6492]: chan_sip.c:4047 retrans_pkt: Retransmission timeout reached on transmission 1342742344_134122685@67.231.5.110 for seqno 591368 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Jul 3 21:05:42] WARNING[6492]: chan_sip.c:4076 retrans_pkt: Hanging up call 1342742344_134122685@67.231.5.110 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
-- Channel SIP/65412378GW1-00000002 left 'native_rtp' basic-bridge <0ee6a0de-c54a-4117-80a4-5d137d97234f>
-- Channel SIP/thufir-00000003 left 'native_rtp' basic-bridge <0ee6a0de-c54a-4117-80a4-5d137d97234f>
Scheduling destruction of SIP dialog '4f4a94fa0f17d1551440124405b35018@192.168.1.8:5060' in 6400 ms (Method: INVITE)
== Spawn extension (inbound, 12345678901, 2) exited non-zero on 'SIP/65412378GW1-00000002'
Scheduling destruction of SIP dialog '1342742344_134122685@67.231.5.110' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.1.5:5062:
BYE sip:thufir@192.168.1.5:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK64a94121;rport
Max-Forwards: 70
From: "PLANET EARTH " <sip:9876543210@192.168.1.8>;tag=as5042a3ac
To: <sip:thufir@192.168.1.5:5062>;tag=441ea2b92b29287ei2
Call-ID: 4f4a94fa0f17d1551440124405b35018@192.168.1.8:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
---
Reliably Transmitting (NAT) to 63.247.69.226:5060:
BYE sip:+19876543210@67.231.5.110:5060 SIP/2.0
Via: SIP/2.0/UDP 123.456.789.123:5060;branch=z9hG4bK0b4f7cc8;rport
Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
Max-Forwards: 70
From: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
To: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
---
<--- SIP read from UDP:192.168.1.5:5062 --->
SIP/2.0 200 OK
To: <sip:thufir@192.168.1.5:5062>;tag=441ea2b92b29287ei2
From: "PLANET EARTH " <sip:9876543210@192.168.1.8>;tag=as5042a3ac
Call-ID: 4f4a94fa0f17d1551440124405b35018@192.168.1.8:5060
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK64a94121
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4f4a94fa0f17d1551440124405b35018@192.168.1.8:5060' Method: INVITE
Retransmitting #1 (NAT) to 63.247.69.226:5060:
BYE sip:+19876543210@67.231.5.110:5060 SIP/2.0
Via: SIP/2.0/UDP 123.456.789.123:5060;branch=z9hG4bK0b4f7cc8;rport
Route: <sip:63.247.69.226;lr=on;ftag=gK08121607;vsf=AAAAAB0BAgcGBgADAgVwAXcYBR0DHQQABB8BMTA-;dlgcor=572.9e41>
Max-Forwards: 70
From: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
To: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
---
<--- SIP read from UDP:63.247.69.226:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.456.789.123:5060;received=123.456.789.123;branch=z9hG4bK0b4f7cc8;rport=49192
From: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
To: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1342742344_134122685@67.231.5.110' Method: INVITE
<--- SIP read from UDP:63.247.69.226:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.456.789.123:5060;received=123.456.789.123;branch=z9hG4bK0b4f7cc8;rport=49192
From: <sip:+12345678901@63.247.69.226>;tag=as0c1d8ac2
To: "PLANET EARTH " <sip:9876543210@67.231.5.110>;tag=gK08121607
Call-ID: 1342742344_134122685@67.231.5.110
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
mordor*CLI>
mordor*CLI>
mordor*CLI> sip set debug off
SIP Debugging Disabled
mordor*CLI>
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