Skip to content

Instantly share code, notes, and snippets.

@elcontrastador

elcontrastador/gist:4b9db410ff9cf14f1029 Secret

Created November 8, 2010 02:09
Show Gist options
  • Save elcontrastador/4b9db410ff9cf14f1029 to your computer and use it in GitHub Desktop.
Save elcontrastador/4b9db410ff9cf14f1029 to your computer and use it in GitHub Desktop.
Download ZIP
Raw
gistfile1.txt
root@mbp-linux:~# asterisk -r
Asterisk 1.6.2.7-1ubuntu1, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.2.7-1ubuntu1 currently running on mbp-linux (pid = 2207)
mbp-linux*CLI> sip set debug on
SIP Debugging re-enabled
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
INVITE sip:5594030000@192.168.1.65 SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKyXm52XFvHD6DS
Max-Forwards: 49
From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=0Bg0B0tQQrtNr
To: <sip:5594030000@192.168.1.65>
Call-ID: febcab06-657f-122e-dfbf-0015c5eaaddb
CSeq: 4255006 INVITE
Contact: <sip:mod_sofia@216.82.225.24:5060>
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: 271
P-Asserted-Identity: "+15596960400" <sip:5596960400@216.82.225.24>
v=0
o=Sonus_UAC 30856 10803 IN IP4 192.168.47.68
s=SIP Media Capabilities
c=IN IP4 67.231.4.98
t=0 0
m=audio 15614 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 - febcab06-657f-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 67.231.4.98:15614
Looking for 5594030000 in default (domain 192.168.1.65)
list_route: hop: <sip:mod_sofia@216.82.225.24:5060>
<--- Transmitting (no NAT) to 216.82.225.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKyXm52XFvHD6DS;received=216.82.225.24;rport=5060
From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=0Bg0B0tQQrtNr
To: <sip:5594030000@192.168.1.65>
Call-ID: febcab06-657f-122e-dfbf-0015c5eaaddb
CSeq: 4255006 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:5594030000@99.28.157.10>
Content-Length: 0
<------------>
Audio is at 99.28.157.10 port 10372
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:5594030000@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK720b4679;rport
Max-Forwards: 70
From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as33783c45
To: <sip:5594030000@trunk1.freepbx.com>
Contact: <sip:100ce039@99.28.157.10>
Call-ID: 6839b6b04e1ffca005fdb22e60933cdd@99.28.157.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no
Date: Mon, 08 Nov 2010 02:09:00 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 1866927827 1866927827 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 10372 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=z9hG4bK720b4679;rport=5060
From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as33783c45
To: <sip:5594030000@trunk1.freepbx.com>
Call-ID: 6839b6b04e1ffca005fdb22e60933cdd@99.28.157.10
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=z9hG4bK720b4679;rport=5060
From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as33783c45
To: <sip:5594030000@trunk1.freepbx.com>;tag=2X2HFpvyHa7tF
Call-ID: 6839b6b04e1ffca005fdb22e60933cdd@99.28.157.10
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="2763e2d4-eadd-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:5594030000@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK720b4679;rport
Max-Forwards: 70
From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as33783c45
To: <sip:5594030000@trunk1.freepbx.com>;tag=2X2HFpvyHa7tF
Contact: <sip:100ce039@99.28.157.10>
Call-ID: 6839b6b04e1ffca005fdb22e60933cdd@99.28.157.10
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no
Content-Length: 0
---
Audio is at 99.28.157.10 port 10372
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:5594030000@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK6e24c68e;rport
Max-Forwards: 70
From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as33783c45
To: <sip:5594030000@trunk1.freepbx.com>
Contact: <sip:100ce039@99.28.157.10>
Call-ID: 6839b6b04e1ffca005fdb22e60933cdd@99.28.157.10
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no
Proxy-Authorization: Digest username="100ce039", realm="trunk1.freepbx.com", algorithm=MD5, uri="sip:5594030000@trunk1.freepbx.com", nonce="2763e2d4-eadd-11df-8944-edffed59eac9", response="439f92e51893ddfba26901ffbef19f8f", qop=auth, cnonce="6c4dbd13", nc=00000001
Date: Mon, 08 Nov 2010 02:09:01 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 1866927827 1866927828 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 10372 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=z9hG4bK6e24c68e;rport=5060
From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as33783c45
To: <sip:5594030000@trunk1.freepbx.com>
Call-ID: 6839b6b04e1ffca005fdb22e60933cdd@99.28.157.10
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:5594030000@192.168.1.65 SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKZ6Dy4r0Zepv0m
Max-Forwards: 69
From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=617mp2Zc6D05D
To: <sip:5594030000@192.168.1.65>
Call-ID: fece831b-657f-122e-dfbf-0015c5eaaddb
CSeq: 4255006 INVITE
Contact: <sip:mod_sofia@216.82.225.24:5060>
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: 260
Remote-Party-ID: "+15596960400" <sip:5596960400@216.82.225.24>;party=calling;screen=no;privacy=off
v=0
o=root 1866927827 1866927828 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 10372 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 - fece831b-657f-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 99.28.157.10:10372
Looking for 5594030000 in default (domain 192.168.1.65)
list_route: hop: <sip:mod_sofia@216.82.225.24:5060>
<--- Transmitting (no NAT) to 216.82.225.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKZ6Dy4r0Zepv0m;received=216.82.225.24;rport=5060
From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=617mp2Zc6D05D
To: <sip:5594030000@192.168.1.65>
Call-ID: fece831b-657f-122e-dfbf-0015c5eaaddb
CSeq: 4255006 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:5594030000@99.28.157.10>
Content-Length: 0
<------------>
Audio is at 99.28.157.10 port 10408
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:5594030000@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK04128335;rport
Max-Forwards: 70
From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as67f100ac
To: <sip:5594030000@trunk1.freepbx.com>
Contact: <sip:100ce039@99.28.157.10>
Call-ID: 6f68c033329a6b4316e15fd52bc1f0f8@99.28.157.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no
Date: Mon, 08 Nov 2010 02:09:01 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 551873170 551873170 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 10408 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=z9hG4bK04128335;rport=5060
From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as67f100ac
To: <sip:5594030000@trunk1.freepbx.com>
Call-ID: 6f68c033329a6b4316e15fd52bc1f0f8@99.28.157.10
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=z9hG4bK04128335;rport=5060
From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as67f100ac
To: <sip:5594030000@trunk1.freepbx.com>;tag=8Kt6Sr1K0ZcBN
Call-ID: 6f68c033329a6b4316e15fd52bc1f0f8@99.28.157.10
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="2775db92-eadd-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:5594030000@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK04128335;rport
Max-Forwards: 70
From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as67f100ac
To: <sip:5594030000@trunk1.freepbx.com>;tag=8Kt6Sr1K0ZcBN
Contact: <sip:100ce039@99.28.157.10>
Call-ID: 6f68c033329a6b4316e15fd52bc1f0f8@99.28.157.10
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no
Content-Length: 0
---
Audio is at 99.28.157.10 port 10408
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:5594030000@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK720f06eb;rport
Max-Forwards: 70
From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as67f100ac
To: <sip:5594030000@trunk1.freepbx.com>
Contact: <sip:100ce039@99.28.157.10>
Call-ID: 6f68c033329a6b4316e15fd52bc1f0f8@99.28.157.10
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;privacy=off;screen=no
Proxy-Authorization: Digest username="100ce039", realm="trunk1.freepbx.com", algorithm=MD5, uri="sip:5594030000@trunk1.freepbx.com", nonce="2775db92-eadd-11df-8944-edffed59eac9", response="b6d77af61afeb1fac3468534f6717b13", qop=auth, cnonce="228989d0", nc=00000001
Date: Mon, 08 Nov 2010 02:09:01 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 551873170 551873171 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 10408 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=z9hG4bK720f06eb;rport=5060
From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as67f100ac
To: <sip:5594030000@trunk1.freepbx.com>
Call-ID: 6f68c033329a6b4316e15fd52bc1f0f8@99.28.157.10
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=z9hG4bK720f06eb;rport=5060
From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as67f100ac
To: <sip:5594030000@trunk1.freepbx.com>;tag=BF6gZ9KyQtF3Q
Call-ID: 6f68c033329a6b4316e15fd52bc1f0f8@99.28.157.10
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:5594030000@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK720f06eb;rport
Max-Forwards: 70
From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as67f100ac
To: <sip:5594030000@trunk1.freepbx.com>;tag=BF6gZ9KyQtF3Q
Contact: <sip:100ce039@99.28.157.10>
Call-ID: 6f68c033329a6b4316e15fd52bc1f0f8@99.28.157.10
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;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=z9hG4bKZ6Dy4r0Zepv0m;received=216.82.225.24;rport=5060
From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=617mp2Zc6D05D
To: <sip:5594030000@192.168.1.65>;tag=as261a88ad
Call-ID: fece831b-657f-122e-dfbf-0015c5eaaddb
CSeq: 4255006 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:5594030000@192.168.1.65 SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKZ6Dy4r0Zepv0m
Max-Forwards: 69
From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=617mp2Zc6D05D
To: <sip:5594030000@192.168.1.65:5060>;tag=as261a88ad
Call-ID: fece831b-657f-122e-dfbf-0015c5eaaddb
CSeq: 4255006 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '6f68c033329a6b4316e15fd52bc1f0f8@99.28.157.10' 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=z9hG4bK6e24c68e;rport=5060
From: "+15596960400" <sip:100ce039@192.168.1.65>;tag=as33783c45
To: <sip:5594030000@trunk1.freepbx.com>;tag=5revm7e9849jj
Call-ID: 6839b6b04e1ffca005fdb22e60933cdd@99.28.157.10
CSeq: 103 INVITE
Contact: <sip:mod_sofia@216.82.225.24:5060;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:5594030000@trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK6e24c68e;rport
Max-Forwards: 70
From: "+15596960400" <sip:100ce039@99.28.157.10>;tag=as33783c45
To: <sip:5594030000@trunk1.freepbx.com>;tag=5revm7e9849jj
Contact: <sip:100ce039@99.28.157.10>
Call-ID: 6839b6b04e1ffca005fdb22e60933cdd@99.28.157.10
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Remote-Party-ID: "+15596960400" <sip:5596960400@99.28.157.10>;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=z9hG4bKyXm52XFvHD6DS;received=216.82.225.24;rport=5060
From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=0Bg0B0tQQrtNr
To: <sip:5594030000@192.168.1.65>;tag=as7daecfce
Call-ID: febcab06-657f-122e-dfbf-0015c5eaaddb
CSeq: 4255006 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:5594030000@192.168.1.65 SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKyXm52XFvHD6DS
Max-Forwards: 49
From: "+15596960400" <sip:5596960400@216.82.225.24>;tag=0Bg0B0tQQrtNr
To: <sip:5594030000@192.168.1.65:5060>;tag=as7daecfce
Call-ID: febcab06-657f-122e-dfbf-0015c5eaaddb
CSeq: 4255006 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '6839b6b04e1ffca005fdb22e60933cdd@99.28.157.10' Method: INVITE
[Nov 7 18:09:07] NOTICE[2241]: chan_sip.c:11459 sip_reregister: -- Re-registration for 100ce039@trunk1.freepbx.com
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 216.82.225.24:5060:
REGISTER sip:trunk1.freepbx.com SIP/2.0
Via: SIP/2.0/UDP 99.28.157.10:5060;branch=z9hG4bK417283e2;rport
Max-Forwards: 70
From: <sip:100ce039@trunk1.freepbx.com>;tag=as4f0b8e24
To: <sip:100ce039@trunk1.freepbx.com>
Call-ID: 440d323827a7d0255e1708581c41f093@192.168.1.65
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1
Authorization: Digest username="100ce039", realm="trunk1.freepbx.com", algorithm=MD5, uri="sip:trunk1.freepbx.com", nonce="ec7a7b10-eadc-11df-8944-edffed59eac9", response="1808acd760b4e718c985e4bea7713e03", qop=auth, cnonce="42e276a8", nc=00000002
Expires: 120
Contact: <sip:100ce039@99.28.157.10>
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=z9hG4bK417283e2;rport=5060
From: <sip:100ce039@trunk1.freepbx.com>;tag=as4f0b8e24
To: <sip:100ce039@trunk1.freepbx.com>;tag=929pgymvjQr3g
Call-ID: 440d323827a7d0255e1708581c41f093@192.168.1.65
CSeq: 104 REGISTER
Contact: <sip:100ce039@99.28.157.10>;expires=120
Date: Mon, 08 Nov 2010 02:09:07 GMT
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Scheduling destruction of SIP dialog '440d323827a7d0255e1708581c41f093@192.168.1.65' in 32000 ms (Method: REGISTER)
[Nov 7 18:09:07] NOTICE[2241]: chan_sip.c:18196 handle_response_register: Outbound Registration: Expiry for trunk1.freepbx.com is 120 sec (Scheduling reregistration in 105 s)
mbp-linux*CLI>
<--- SIP read from UDP:216.82.225.24:5060 --->
NOTIFY sip:100ce039@192.168.1.65 SIP/2.0
Via: SIP/2.0/UDP 216.82.225.24;rport;branch=z9hG4bKeFjBrgr56rKFN
Max-Forwards: 70
From: <sip:100ce039@trunk1.freepbx.com>;tag=BNv8Kmp3c948Q
To: <sip:100ce039@trunk1.freepbx.com>
Call-ID: 02ab1ef0-6580-122e-dfbf-0015c5eaaddb
CSeq: 4255009 NOTIFY
Contact: <sip:mod_sofia@216.82.225.24:5060>
User-Agent: FreePBX Trunking
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Event: message-summary
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Subscription-State: terminated;reason=timeout
Content-Type: application/simple-message-summary
Content-Length: 74
Messages-Waiting: no
Message-Account: sip:100ce039@trunk1.freepbx.com
<------------->
--- (16 headers 3 lines) ---
<--- Transmitting (NAT) to 216.82.225.24:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 216.82.225.24;branch=z9hG4bKeFjBrgr56rKFN;received=216.82.225.24;rport=5060
From: <sip:100ce039@trunk1.freepbx.com>;tag=BNv8Kmp3c948Q
To: <sip:100ce039@trunk1.freepbx.com>;tag=as6a828ca0
Call-ID: 02ab1ef0-6580-122e-dfbf-0015c5eaaddb
CSeq: 4255009 NOTIFY
Server: Asterisk PBX 1.6.2.7-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
Sign up for free to join this conversation on GitHub. Already have an account? Sign in to comment