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Created November 8, 2010 01:42
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gistfile1.txt
mbp-linux*CLI> sip set debug on
SIP Debugging enabled
mbp-linux*CLI>
<--- SIP read from UDP:192.168.1.64:43533 --->
<------------->
mbp-linux*CLI>
<--- SIP read from UDP:192.168.1.64:59247 --->time
<------------->
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKDpU80pZmpgm4c
Max-Forwards: 49
From: "+15596960400" <sip:[email protected]>;tag=cpS1crj11HatQ
To: <sip:[email protected]>
Call-ID: 292f1216-657b-122e-dfbf-0015c5eaaddb
CSeq: 4253968 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
P-Asserted-Identity: "+15596960400" <sip:[email protected]>
v=0
o=Sonus_UAC 7878 27307 IN IP4 192.168.47.68
s=SIP Media Capabilities
c=IN IP4 67.231.4.99
t=0 0
m=audio 14988 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=maxptime:20
<------------->
--- (17 headers 12 lines) ---
Sending to 216.82.225.24 : 5060 (NAT)
Using INVITE request as basis request - 292f1216-657b-122e-dfbf-0015c5eaaddb
Found peer 'ext-sip-account' for '5596960400' from 216.82.225.24:5060
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 - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=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 67.231.4.99:14988
Looking for 5594030000 in default (domain 192.168.1.65)
list_route: hop: <sip:[email protected]:5060>
<--- Transmitting (no NAT) to 216.82.225.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKDpU80pZmpgm4c;received=216.82.225.24;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=cpS1crj11HatQ
To: <sip:[email protected]>
Call-ID: 292f1216-657b-122e-dfbf-0015c5eaaddb
CSeq: 4253968 INVITE
Server: Asterisk PBX 1.6.2.7-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
Audio is at 99.28.157.10 port 19242
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 216.82.225.24:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK1a40b322;rport
Max-Forwards: 70
From: "+15596960400" <sip:[email protected]>;tag=as662ebece
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:[email protected]>;privacy=off;screen=no
Date: Mon, 08 Nov 2010 01:34:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 261348102 261348102 IN IP4 99.28.157.10
s=Asterisk PBX 1.6.2.7-1ubuntu1
c=IN IP4 99.28.157.10
t=0 0
m=audio 19242 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
---
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK1a40b322;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=as662ebece
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FreePBX Trunking
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK1a40b322;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=as662ebece
To: <sip:[email protected]>;tag=DZjteK34yt0cK
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FreePBX Trunking
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm="trunk1.freepbx.com", nonce="51d6a718-ead8-11df-8944-edffed59eac9", algorithm=MD5, qop="auth"
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Transmitting (no NAT) to 216.82.225.24:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK1a40b322;rport
Max-Forwards: 70
From: "+15596960400" <sip:[email protected]>;tag=as662ebece
To: <sip:[email protected]>;tag=DZjteK34yt0cK
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:[email protected]>;privacy=off;screen=no
Content-Length: 0
---
Audio is at 99.28.157.10 port 19242
Adding codec 0x4 (ulaw) to SDP
dding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 216.82.225.24:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK31ff4f96;rport
Max-Forwards: 70
From: "+15596960400" <sip:[email protected]>;tag=as662ebece
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:[email protected]>;privacy=off;screen=no
Proxy-Authorization: Digest username="100ce039", realm="trunk1.freepbx.com", algorithm=MD5, uri="sip:[email protected]", nonce="51d6a718-ead8-11df-8944-edffed59eac9", response="6a2d195a824230445425db4cbd11ba5a", qop=auth, cnonce="5b001aea", nc=00000001
Date: Mon, 08 Nov 2010 01:34:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 261348102 261348103 IN IP4 99.28.157.10
s=Asterisk PBX 1.6.2.7-1ubuntu1
c=IN IP4 99.28.157.10
t=0 0
m=audio 19242 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
---
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK31ff4f96;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=as662ebece
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: FreePBX Trunking
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKeZm12HgrKSaQr
Max-Forwards: 69
From: "+15596960400" <sip:[email protected]>;tag=FH5Bj94BScDja
To: <sip:[email protected]>
Call-ID: 2941d1a7-657b-122e-dfbf-0015c5eaaddb
CSeq: 4253968 INVITE
Contact: <sip:[email protected]:5060>
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 258
Remote-Party-ID: "+15596960400" <sip:[email protected]>;party=calling;screen=no;privacy=off
v=0
o=root 261348102 261348103 IN IP4 99.28.157.10
s=Asterisk PBX 1.6.2.7-1ubuntu1
c=IN IP4 99.28.157.10
t=0 0
m=audio 19242 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
<------------->
--- (16 headers 11 lines) ---
Sending to 216.82.225.24 : 5060 (NAT)
Using INVITE request as basis request - 2941d1a7-657b-122e-dfbf-0015c5eaaddb
Found peer 'ext-sip-account' for '5596960400' from 216.82.225.24:5060
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 - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=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 99.28.157.10:19242
Looking for 5594030000 in default (domain 192.168.1.65)
list_route: hop: <sip:[email protected]:5060>
<--- Transmitting (no NAT) to 216.82.225.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKeZm12HgrKSaQr;received=216.82.225.24;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=FH5Bj94BScDja
To: <sip:[email protected]>
Call-ID: 2941d1a7-657b-122e-dfbf-0015c5eaaddb
CSeq: 4253968 INVITE
Server: Asterisk PBX 1.6.2.7-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
Audio is at 99.28.157.10 port 16054
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 216.82.225.24:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK34442773;rport
Max-Forwards: 70
From: "+15596960400" <sip:[email protected]>;tag=as5c4b531a
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:[email protected]>;privacy=off;screen=no
Date: Mon, 08 Nov 2010 01:34:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 272
v=0
o=root 1191172620 1191172620 IN IP4 99.28.157.10
s=Asterisk PBX 1.6.2.7-1ubuntu1
c=IN IP4 99.28.157.10
t=0 0
m=audio 16054 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
---
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK34442773;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=as5c4b531a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FreePBX Trunking
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK34442773;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=as5c4b531a
To: <sip:[email protected]>;tag=gty4K4NFpN34N
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FreePBX Trunking
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm="trunk1.freepbx.com", nonce="51e923d4-ead8-11df-8944-edffed59eac9", algorithm=MD5, qop="auth"
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Transmitting (no NAT) to 216.82.225.24:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK34442773;rport
Max-Forwards: 70
From: "+15596960400" <sip:[email protected]>;tag=as5c4b531a
To: <sip:[email protected]>;tag=gty4K4NFpN34N
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:[email protected]>;privacy=off;screen=no
Content-Length: 0
---
Audio is at 99.28.157.10 port 16054
Adding codec 0x4 (ulaw) to SDP
dding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 216.82.225.24:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK6eda181a;rport
Max-Forwards: 70
From: "+15596960400" <sip:[email protected]>;tag=as5c4b531a
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:[email protected]>;privacy=off;screen=no
Proxy-Authorization: Digest username="100ce039", realm="trunk1.freepbx.com", algorithm=MD5, uri="sip:[email protected]", nonce="51e923d4-ead8-11df-8944-edffed59eac9", response="64d622927159ddfe8fe699044db4fba2", qop=auth, cnonce="049158b8", nc=00000001
Date: Mon, 08 Nov 2010 01:34:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 272
v=0
o=root 1191172620 1191172621 IN IP4 99.28.157.10
s=Asterisk PBX 1.6.2.7-1ubuntu1
c=IN IP4 99.28.157.10
t=0 0
m=audio 16054 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
---
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK6eda181a;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=as5c4b531a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: FreePBX Trunking
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK6eda181a;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=as5c4b531a
To: <sip:[email protected]>;tag=KNaFSN8SDg6vr
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: FreePBX Trunking
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Transmitting (no NAT) to 216.82.225.24:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK6eda181a;rport
Max-Forwards: 70
From: "+15596960400" <sip:[email protected]>;tag=as5c4b531a
To: <sip:[email protected]>;tag=KNaFSN8SDg6vr
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:[email protected]>;privacy=off;screen=no
Content-Length: 0
---
<--- Reliably Transmitting (no NAT) to 216.82.225.24:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKeZm12HgrKSaQr;received=216.82.225.24;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=FH5Bj94BScDja
To: <sip:[email protected]>;tag=as08867e89
Call-ID: 2941d1a7-657b-122e-dfbf-0015c5eaaddb
CSeq: 4253968 INVITE
Server: Asterisk PBX 1.6.2.7-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Content-Length: 0
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
<------------>
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKeZm12HgrKSaQr
Max-Forwards: 69
From: "+15596960400" <sip:[email protected]>;tag=FH5Bj94BScDja
To: <sip:[email protected]:5060>;tag=as08867e89
Call-ID: 2941d1a7-657b-122e-dfbf-0015c5eaaddb
CSeq: 4253968 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK31ff4f96;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=as662ebece
To: <sip:[email protected]>;tag=e8BKgem8U3pZe
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected]:5060;transport=udp>
User-Agent: FreePBX Trunking
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Transmitting (no NAT) to 216.82.225.24:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK31ff4f96;rport
Max-Forwards: 70
From: "+15596960400" <sip:[email protected]>;tag=as662ebece
To: <sip:[email protected]>;tag=e8BKgem8U3pZe
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:[email protected]>;privacy=off;screen=no
Content-Length: 0
---
<--- Reliably Transmitting (no NAT) to 216.82.225.24:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKDpU80pZmpgm4c;received=216.82.225.24;rport=5060
From: "+15596960400" <sip:[email protected]>;tag=cpS1crj11HatQ
To: <sip:[email protected]>;tag=as75eb340a
Call-ID: 292f1216-657b-122e-dfbf-0015c5eaaddb
CSeq: 4253968 INVITE
Server: Asterisk PBX 1.6.2.7-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Content-Length: 0
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
<------------>
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKDpU80pZmpgm4c
Max-Forwards: 49
From: "+15596960400" <sip:[email protected]>;tag=cpS1crj11HatQ
To: <sip:[email protected]:5060>;tag=as75eb340a
Call-ID: 292f1216-657b-122e-dfbf-0015c5eaaddb
CSeq: 4253968 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
Really destroying SIP dialog '[email protected]' Method: REGISTER
Reliably Transmitting (no NAT) to 216.82.225.24:5060:
OPTIONS sip:trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK52cfa45e;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as1880ca6b
To: <sip:trunk1.freepbx.com>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Date: Mon, 08 Nov 2010 01:34:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.65:5060;branch=z9hG4bK52cfa45e;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as1880ca6b
To: <sip:trunk1.freepbx.com>;tag=g95Z7yBcBa9je
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: <sip:216.82.225.24>
User-Agent: FreePBX Trunking
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS
mbp-linux*CLI>
<--- SIP read from UDP:192.168.1.64:43533 --->
<------------->
mbp-linux*CLI>
<--- SIP read from UDP:192.168.1.64:59247 --->
<------------->
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