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#724093
sip.conf
; AstBill-0.9.0.8
; Created: Mon Oct 24 16:47:33 BST 2005
; sip_custom.conf
; This is where you put your own STATIC custom SIP configuration
;
; Remember Asterisk commands: reload or extensions reload
[70555]
type=peer
username=70555
fromuser=70555
context=default
secret=65703
host=83.143.155.28
;callerid=Barbara <70125>
;context=default
dtmfmode=rfc2833
;mailbox=70125
[linksys]
context=office
type=friend
username=111
secret=111
host=192.168.5.24
dtmfmode=rfc2833
;disallow=all
;allow=alaw
qualify=yes
[222]
type=friend
username=222
secret=222
callerid=222
host=dynamic
context=office
dtmfmode=rfc2833
;disallow=all
;allow=alaw
qualify=yes
extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. Included contexts are included in the order
; they are listed.
;
[default]
exten => #,1,Hangup ; Hang them up.
; A timeout and "invalid extension rule"
exten => t,1,Goto(i,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
exten => i,2,Hangup ; Hang them up.
exten => h,1,DeadAGI(agistardead.agi)
exten => h,1,Hangup
exten => 198,1,Ringing
exten => 198,2,Wait(2)
exten => 198,3,VoicemailMain
exten => 199,1,Ringing
exten => 199,2,Wait(2)
exten => 199,3,AGI(agistarvoicemail.agi)
exten => 199,4,VoicemailMain(s${VOICEMAIL})
exten => 500,1,Ringing
exten => 500,2,Wait(2)
exten => 500,3,DateTime
exten => 500,4,Hangup ; Hang them up.
*CLI> sip debug
SIP Debugging enabled
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
*CLI>
<--- SIP read from 192.168.5.1:5060 --->
INVITE sip:421818@192.168.5.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK37537693b718f1e1
Max-Forwards: 70
To: <sip:421818@192.168.5.100>
From: <sip:222@192.168.5.100;user=phone>;tag=xIzN3EDMxID
Call-ID: 243D1B916121212@192.168.5.1
CSeq: 1 INVITE
Contact: <sip:222@192.168.5.1:5060>
Session-Expires: 180;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Length: 234
v=0
o=222 0 0 IN IP4 192.168.5.1
s=-
c=IN IP4 192.168.5.1
t=0 0
m=audio 50364 RTP/AVP 0 18 100
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:18 G729/8000
a=sendrecv
a=ptime:20
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
<------------->
--- (12 headers 13 lines) ---
Sending to 192.168.5.1 : 5060 (no NAT)
Using INVITE request as basis request - 243D1B916121212@192.168.5.1
<--- Reliably Transmitting (no NAT) to 192.168.5.1:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK37537693b718f1e1;received=192.168.5.1
From: <sip:222@192.168.5.100;user=phone>;tag=xIzN3EDMxID
To: <sip:421818@192.168.5.100>;tag=as2b2032ac
Call-ID: 243D1B916121212@192.168.5.1
CSeq: 1 INVITE
User-Agent: Noname
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="017ee81a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '243D1B916121212@192.168.5.1' in 32000 ms (Method: INVITE)
Found user '222'
<--- SIP read from 192.168.5.1:5060 --->
ACK sip:421818@192.168.5.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK37537693b718f1e1
Max-Forwards: 70
To: <sip:421818@192.168.5.100>;tag=as2b2032ac
From: <sip:222@192.168.5.100;user=phone>;tag=xIzN3EDMxID
Call-ID: 243D1B916121212@192.168.5.1
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 192.168.5.1:5060 --->
INVITE sip:421818@192.168.5.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bKe00036c520796f26
Max-Forwards: 70
To: <sip:421818@192.168.5.100>
From: <sip:222@192.168.5.100;user=phone>;tag=xIzN3EDMxID
Call-ID: 243D1B916121212@192.168.5.1
CSeq: 2 INVITE
Contact: <sip:222@192.168.5.1:5060>
Session-Expires: 180;refresher=uac
Supported: timer
Proxy-Authorization: Digest username="222", realm="asterisk", nonce="017ee81a", uri="sip:421818@192.168.5.100", response="b33f1a33af044df4210111e68b360d53", algorithm=MD5, nc=0000001
Content-Type: application/sdp
Content-Length: 234
v=0
o=222 0 0 IN IP4 192.168.5.1
s=-
c=IN IP4 192.168.5.1
t=0 0
m=audio 50364 RTP/AVP 0 18 100
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:18 G729/8000
a=sendrecv
a=ptime:20
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
<------------->
--- (13 headers 13 lines) ---
Sending to 192.168.5.1 : 5060 (no NAT)
Using INVITE request as basis request - 243D1B916121212@192.168.5.1
Found user '222'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 100
Peer audio RTP is at port 192.168.5.1:50364
Found description format PCMU for ID 0
Found description format G729 for ID 18
Found description format telephone-event for ID 100
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x104 (ulaw|g729)/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 192.168.5.1:50364
Looking for 421818 in office (domain 192.168.5.100)
list_route: hop: <sip:222@192.168.5.1:5060>
<--- Transmitting (no NAT) to 192.168.5.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bKe00036c520796f26;received=192.168.5.1
From: <sip:222@192.168.5.100;user=phone>;tag=xIzN3EDMxID
To: <sip:421818@192.168.5.100>
Call-ID: 243D1B916121212@192.168.5.1
CSeq: 2 INVITE
User-Agent: Noname
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:421818@192.168.5.100>
Content-Length: 0
<------------>
-- Executing [421818@office:1] Dial("SIP/222-081fa5b0", "SIP/421818@linksys|20") in new stack
Audio is at 192.168.5.100 port 10554
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.5.24:5060:
INVITE sip:421818@192.168.5.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK1c3c210b;rport
From: "222" <sip:222@192.168.5.100>;tag=as0387a7e6
To: <sip:421818@192.168.5.24>
Contact: <sip:222@192.168.5.100>
Call-ID: 68a154c6376ccb1126896235676c7712@192.168.5.100
CSeq: 102 INVITE
User-Agent: Noname
Max-Forwards: 70
Date: Wed, 03 Oct 2007 08:53:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 17570 17570 IN IP4 192.168.5.100
s=session
c=IN IP4 192.168.5.100
t=0 0
m=audio 10554 RTP/AVP 0 3 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 421818@linksys
<--- SIP read from 192.168.5.24:5060 --->
SIP/2.0 404 Not Found
To: <sip:421818@192.168.5.24>;tag=be972e5e3c29a9a0i0
From: "222" <sip:222@192.168.5.100>;tag=as0387a7e6
Call-ID: 68a154c6376ccb1126896235676c7712@192.168.5.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK1c3c210b
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.5.24:5060:
ACK sip:421818@192.168.5.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK1c3c210b;rport
From: "222" <sip:222@192.168.5.100>;tag=as0387a7e6
To: <sip:421818@192.168.5.24>;tag=be972e5e3c29a9a0i0
Contact: <sip:222@192.168.5.100>
Call-ID: 68a154c6376ccb1126896235676c7712@192.168.5.100
CSeq: 102 ACK
User-Agent: Noname
Max-Forwards: 70
Content-Length: 0
---
-- SIP/linksys-08201780 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [421818@office:2] Congestion("SIP/222-081fa5b0", "") in new stack
<--- Transmitting (no NAT) to 192.168.5.1:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bKe00036c520796f26;received=192.168.5.1
From: <sip:222@192.168.5.100;user=phone>;tag=xIzN3EDMxID
To: <sip:421818@192.168.5.100>;tag=as4b788a4f
Call-ID: 243D1B916121212@192.168.5.1
CSeq: 2 INVITE
User-Agent: Noname
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:421818@192.168.5.100>
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
<------------>
== Spawn extension (office, 421818, 2) exited non-zero on 'SIP/222-081fa5b0'
<--- SIP read from 192.168.5.1:5060 --->
ACK sip:421818@192.168.5.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bKe00036c520796f26
Max-Forwards: 70
To: <sip:421818@192.168.5.100>;tag=as4b788a4f
From: <sip:222@192.168.5.100;user=phone>;tag=xIzN3EDMxID
Call-ID: 243D1B916121212@192.168.5.1
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '68a154c6376ccb1126896235676c7712@192.168.5.100' Method: INVITE
Really destroying SIP dialog '243D1B916121212@192.168.5.1' Method: ACK
Reliably Transmitting (no NAT) to 192.168.5.24:5060:
OPTIONS sip:192.168.5.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK307c546f;rport
From: "asterisk" <sip:asterisk@192.168.5.100>;tag=as09b8ae40
To: <sip:192.168.5.24>
Contact: <sip:asterisk@192.168.5.100>
Call-ID: 19a7fdd35da6d38b3b75d7c0683fec6e@192.168.5.100
CSeq: 102 OPTIONS
User-Agent: Noname
Max-Forwards: 70
Date: Wed, 03 Oct 2007 08:53:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
Reliably Transmitting (no NAT) to 192.168.5.1:5060:
OPTIONS sip:222@192.168.5.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK5a09b74f;rport
From: "asterisk" <sip:asterisk@192.168.5.100>;tag=as504711e1
To: <sip:222@192.168.5.1:5060>
Contact: <sip:asterisk@192.168.5.100>
Call-ID: 7ae227663b122c0d19488512369c7ee8@192.168.5.100
CSeq: 102 OPTIONS
User-Agent: Noname
Max-Forwards: 70
Date: Wed, 03 Oct 2007 08:53:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 192.168.5.24:5060 --->
SIP/2.0 404 Not Found
To: <sip:192.168.5.24>;tag=be972e5e3c29a9a0i0
From: "asterisk" <sip:asterisk@192.168.5.100>;tag=as09b8ae40
Call-ID: 19a7fdd35da6d38b3b75d7c0683fec6e@192.168.5.100
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK307c546f
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '19a7fdd35da6d38b3b75d7c0683fec6e@192.168.5.100' Method: OPTIONS
<--- SIP read from 192.168.5.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK5a09b74f
To: <sip:222@192.168.5.1:5060>
From: "asterisk" <sip:asterisk@192.168.5.100>;tag=as504711e1
Call-ID: 7ae227663b122c0d19488512369c7ee8@192.168.5.100
CSeq: 102 OPTIONS
Supported: timer,100rel
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL
Contact: <sip:222@192.168.5.1:5060>
Content-Length: 0
Accept-Encoding: identity
Accept-Language: en
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '7ae227663b122c0d19488512369c7ee8@192.168.5.100' Method: OPTIONS