All pastes #565028 Raw Edit

Anonymous

public text v1 · immutable
·published 2007-06-13 20:50 UTC
rendered paste body
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.0.4.3:5060;branch=z9hG4bK50383360;rport=5060
From: "asterisk" <sip:asterisk@10.0.4.3>;tag=as42cadd58
To: <sip:200@10.0.249.254:2054;line=dmwk91jz>
Call-ID: 044beff041fecf6d7eca64aa7b59f6c2@10.0.4.3
CSeq: 102 NOTIFY
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '044beff041fecf6d7eca64aa7b59f6c2@10.0.4.3' Method: NOTIFY
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
INVITE sip:16175043652@10.0.4.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-x65qzoocqrd1;rport
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:200@10.0.249.254:2054;line=dmwk91jz>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.8
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 1281895614 1281895614 IN IP4 10.0.249.254
s=call
c=IN IP4 10.0.249.254
t=0 0
m=audio 51846 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0XTYHWud6CX7Y8jkdNMtZlQTMXg3WO7Nkowm/60j
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

<------------->
--- (18 headers 19 lines) ---
Sending to 10.0.249.254 : 2054 (NAT)
Using INVITE request as basis request - 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
Found peer '200'

<--- Reliably Transmitting (no NAT) to 10.0.249.254:2054 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-x65qzoocqrd1;received=10.0.249.254;rport=2054
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as62813a25
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02e143d1"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c643582dbba-rgbkvqf3mple@snom360-00041323719F' in 32000 ms (Method: INVITE)
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
ACK sip:16175043652@10.0.4.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-x65qzoocqrd1;rport
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as62813a25
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:200@10.0.249.254:2054;line=dmwk91jz>;flow-id=1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
INVITE sip:16175043652@10.0.4.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-zvpso6od9cuf;rport
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:200@10.0.249.254:2054;line=dmwk91jz>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.8
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username="200",realm="asterisk",nonce="02e143d1",uri="sip:16175043652@10.0.4.3;user=phone",response="74b4594598cbba4f4f4147f17bd5dee7",algorithm=md5
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 1281895614 1281895614 IN IP4 10.0.249.254
s=call
c=IN IP4 10.0.249.254
t=0 0
m=audio 51846 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0XTYHWud6CX7Y8jkdNMtZlQTMXg3WO7Nkowm/60j
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

<------------->
--- (19 headers 19 lines) ---
Sending to 10.0.249.254 : 2054 (NAT)
Using INVITE request as basis request - 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
Found peer '200'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 10.0.249.254:51846
Found description format pcmu for ID 0
Found description format pcma for ID 8
Found description format g722 for ID 9
Found description format g726-32 for ID 2
Found description format gsm for ID 3
Found description format g729 for ID 18
Found description format g723 for ID 4
Found description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/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 10.0.249.254:51846
Looking for 16175043652 in numberplan-custom-1 (domain 10.0.4.3)
list_route: hop: <sip:200@10.0.249.254:2054;line=dmwk91jz>

<--- Transmitting (no NAT) to 10.0.249.254:2054 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-zvpso6od9cuf;received=10.0.249.254;rport=2054
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:16175043652@10.0.4.3>
Content-Length: 0


<------------>
    -- Executing [16175043652@numberplan-custom-1:1] Macro("SIP/200-00745370", "trunkdial|SIP/trunk_1/16175043652") in new stack
    -- Executing [s@macro-trunkdial:1] Dial("SIP/200-00745370", "SIP/trunk_1/16175043652") in new stack
    -- Called trunk_1/16175043652
    -- Call on SIP/trunk_1-00766f50 left from hold
    -- SIP/trunk_1-00766f50 is making progress passing it to SIP/200-00745370
Audio is at 10.0.4.3 port 18272
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 10.0.249.254:2054 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-zvpso6od9cuf;received=10.0.249.254;rport=2054
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as7ea30464
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:16175043652@10.0.4.3>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 3216 3216 IN IP4 10.0.4.3
s=session
c=IN IP4 10.0.4.3
t=0 0
m=audio 18272 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

<------------>
    -- Call on SIP/trunk_1-00766f50 left from hold
    -- SIP/trunk_1-00766f50 answered SIP/200-00745370
Audio is at 10.0.4.3 port 18272
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.0.249.254:2054 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-zvpso6od9cuf;received=10.0.249.254;rport=2054
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as7ea30464
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:16175043652@10.0.4.3>
Content-Type: application/sdp
Content-Length: 230

v=0
o=root 3216 3217 IN IP4 10.0.4.3
s=session
c=IN IP4 10.0.4.3
t=0 0
m=audio 18272 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

<------------>
    -- Packet2Packet bridging SIP/200-00745370 and SIP/trunk_1-00766f50
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
ACK sip:16175043652@10.0.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-00q2sszyhpi6;rport
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as7ea30464
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:200@10.0.249.254:2054;line=dmwk91jz>;flow-id=1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
BYE sip:16175043652@10.0.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7rlxtyasgr1y;rport
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as7ea30464
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:200@10.0.249.254:2054;line=dmwk91jz>;flow-id=1
User-Agent: snom360/6.5.8
RTP-RxStat: Total_Rx_Pkts=299,Rx_Pkts=299,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=547,Tx_Pkts=547,Remote_Tx_Pkts=297
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 10.0.249.254 : 2054 (NAT)

<--- Transmitting (NAT) to 10.0.249.254:2054 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7rlxtyasgr1y;received=10.0.249.254;rport=2054
From: "BTC" <sip:200@10.0.4.3>;tag=4sp9m5wwkm
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as7ea30464
Call-ID: 3c643582dbba-rgbkvqf3mple@snom360-00041323719F
CSeq: 3 BYE
ser-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:16175043652@10.0.4.3>
Content-Length: 0


<------------>
  == Spawn extension (macro-trunkdial, s, 1) exited non-zero on 'SIP/200-00745370' in macro 'trunkdial'
  == Spawn extension (macro-trunkdial, s, 1) exited non-zero on 'SIP/200-00745370'
Really destroying SIP dialog '3c643582dbba-rgbkvqf3mple@snom360-00041323719F' Method: BYE
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
INVITE sip:16175043652@10.0.4.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7gfpqr1tmgde;rport
From: "BTC" <sip:200@10.0.4.3>;tag=9xwftzfxej
To: <sip:16175043652@10.0.4.3;user=phone>
Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:200@10.0.249.254:2054;line=dmwk91jz>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.8
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 449870880 449870880 IN IP4 10.0.249.254
s=call
c=IN IP4 10.0.249.254
t=0 0
m=audio 49762 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aF3cZLtxa7hvj2dP6JZXgnz0/HJ9NfXNu8En92/K
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

<------------->
--- (18 headers 19 lines) ---
Sending to 10.0.249.254 : 2054 (NAT)
Using INVITE request as basis request - 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F
Found peer '200'

<--- Reliably Transmitting (no NAT) to 10.0.249.254:2054 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7gfpqr1tmgde;received=10.0.249.254;rport=2054
From: "BTC" <sip:200@10.0.4.3>;tag=9xwftzfxej
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as24d8bfd6
Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a74886a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3c6435a130d4-uyv1juesmkdd@snom360-00041323719F' in 32000 ms (Method: INVITE)
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
ACK sip:16175043652@10.0.4.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-7gfpqr1tmgde;rport
From: "BTC" <sip:200@10.0.4.3>;tag=9xwftzfxej
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as24d8bfd6
Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:200@10.0.249.254:2054;line=dmwk91jz>;flow-id=1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
INVITE sip:16175043652@10.0.4.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-w9kwf6sh9i9m;rport
From: "BTC" <sip:200@10.0.4.3>;tag=9xwftzfxej
To: <sip:16175043652@10.0.4.3;user=phone>
Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:200@10.0.249.254:2054;line=dmwk91jz>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.8
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username="200",realm="asterisk",nonce="0a74886a",uri="sip:16175043652@10.0.4.3;user=phone",response="585b1b9653c3378602b1545b87311b9d",algorithm=md5
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 449870880 449870880 IN IP4 10.0.249.254
s=call
c=IN IP4 10.0.249.254
t=0 0
m=audio 49762 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aF3cZLtxa7hvj2dP6JZXgnz0/HJ9NfXNu8En92/K
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

<------------->
--- (19 headers 19 lines) ---
Sending to 10.0.249.254 : 2054 (NAT)
Using INVITE request as basis request - 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F
Found peer '200'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 10.0.249.254:49762
Found description format pcmu for ID 0
Found description format pcma for ID 8
Found description format g722 for ID 9
Found description format g726-32 for ID 2
Found description format gsm for ID 3
Found description format g729 for ID 18
Found description format g723 for ID 4
Found description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/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 10.0.249.254:49762
Looking for 16175043652 in numberplan-custom-1 (domain 10.0.4.3)
list_route: hop: <sip:200@10.0.249.254:2054;line=dmwk91jz>

<--- Transmitting (no NAT) to 10.0.249.254:2054 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-w9kwf6sh9i9m;received=10.0.249.254;rport=2054
From: "BTC" <sip:200@10.0.4.3>;tag=9xwftzfxej
To: <sip:16175043652@10.0.4.3;user=phone>
Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:16175043652@10.0.4.3>
Content-Length: 0


<------------>
    -- Executing [16175043652@numberplan-custom-1:1] Macro("SIP/200-00753790", "trunkdial|SIP/trunk_1/16175043652") in new stack
    -- Executing [s@macro-trunkdial:1] Dial("SIP/200-00753790", "SIP/trunk_1/16175043652") in new stack
    -- Called trunk_1/16175043652
    -- SIP/trunk_1-00743740 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-trunkdial:2] Goto("SIP/200-00753790", "s-CONGESTION|1") in new stack
    -- Goto (macro-trunkdial,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-trunkdial:1] NoOp("SIP/200-00753790", "") in new stack
  == Auto fallthrough, channel 'SIP/200-00753790' status is 'CONGESTION'
pbx*CLI> 
<--- Transmitting (no NAT) to 10.0.249.254:2054 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-w9kwf6sh9i9m;received=10.0.249.254;rport=2054
From: "BTC" <sip:200@10.0.4.3>;tag=9xwftzfxej
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as3159d4b5
Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:16175043652@10.0.4.3>
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1


<------------>
pbx*CLI> 
<--- SIP read from 10.0.249.254:2054 --->
ACK sip:16175043652@10.0.4.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.249.254:2054;branch=z9hG4bK-w9kwf6sh9i9m;rport
From: "BTC" <sip:200@10.0.4.3>;tag=9xwftzfxej
To: <sip:16175043652@10.0.4.3;user=phone>;tag=as3159d4b5
Call-ID: 3c6435a130d4-uyv1juesmkdd@snom360-00041323719F
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:200@10.0.249.254:2054;line=dmwk91jz>;flow-id=1
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '3c6435a130d4-uyv1juesmkdd@snom360-00041323719F' Method: ACK
[Jun 13 16:44:26] NOTICE[3267]: chan_sip.c:7147 sip_reregister:    -- Re-registration for  prichar@inbound1.vitelity.net
[Jun 13 16:44:26] NOTICE[3267]: chan_sip.c:12117 handle_response_register: Outbound Registration: Expiry for inbound1.vitelity.net is 60 sec (Scheduling reregistration in 45 s)