Abandoned internal calls ring for several minutes

I have Asterisk 1.2.13 compiled from scratch running on CentOS 4.2. I also installed the Asterisk sample files. I have only changed the sip and extensions files and all others are the stock samples. I haven’t configured any trunks yet.

I can successfully complete internal calls between the Cisco 7970/SIP phones and/or the X-Lite softphones. The problem is that when an internal call is placed from any phone to one of the Cisco phones and is then abandonded, the dialed phone will continue to ring for several minutes. Stopping and starting Asterisk doesn’t stop the ringing. Neither does a “restart now”.

When I previously used Trixbox with the same phones, I did not see this behavior. Also, when the X-Lite softphone is dialed and the call is abandonded, it works as expected - the softphone immediately quits ringing.

Any ideas why this is occuring or how to fix it?

Thanks in advance!

SIP.CONF:

[ken]
type=friend
username=ken
secret=secret
host=dynamic
context=default

[bill]
type=friend
username=bill
secret=secret
host=dynamic
context=default

[pat]
type=friend
username=pat
secret=secret
host=dynamic
context=default

[x-lite]
type=friend
username=x-lite
secret=secret
host=dynamic
context=default

EXTENSIONS.CONF:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)

[dundi-e164-canonical]

[dundi-e164-customers]

[dundi-e164-via-pstn]

[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
switch => DUNDi/e164

[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch

[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)

[iaxprovider]

[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkld]
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunklocal]
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunktollfree]
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[international]
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
ignorepat => 9
include => local
include => trunkld

[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

[macro-stdexten];
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start

exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain

[macro-stdPrivacyexten];
exten => s,1,Dial(${ARG2},20|p) ; Ring the interface, 20 seconds maximum, call screening option (or use P for databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start

exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start

exten => s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to send this call to a polite “Don’t call again” script.

exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this call to a telemarketer torture script.

exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain

[demo]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.

exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the congratulations

exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; “Please hold while…”
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})

exten => 1235,1,Voicemail(u1234) ; Right to voicemail

exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234) ; Unless busy

;
; # for when they’re done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.

;
; A timeout and “invalid extension rule”
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; “That’s not valid, try again”

;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what’s going on
exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn’t connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.

;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what’s going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it’s over
exten => 600,n,Goto(s,6) ; Start over

;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
; Here’s what a phone entry would look like (IXJ for example)
;

[default]
include => demo

exten => 101,1,Dial(SIP/ken, 10)
exten => 102,1,Dial(SIP/bill, 10)
exten => 104,1,Dial(SIP/pat, 10)
exten => 109,1,Dial(SIP/x-lite, 10)

My guess is that the locanet setting is wrong.

Of course seeing the debug of a call would help a bit

Ian:

Thanks for the response. I appreciate any insight you can provide.

==================
KEN CALLS PAT
(Both Cisco 7970 phones)

*CLI> [Nov 19 07:50:51] – Executing Dial(“SIP/ken-0982b738”, “SIP/pat| 10”) in new stack
[Nov 19 07:50:51] – Called pat
[Nov 19 07:51:01] – Nobody picked up in 10000 ms
[Nov 19 07:51:01] == Auto fallthrough, channel ‘SIP/ken-0982b738’ status is 'NOANSWER’
NOTE: Dialed phone continues to ring for several minutes…

==================
KEN CALLS X-LITE
(Cisco 7970 => X-LITE Softphone)

[Nov 19 07:56:48] – Executing Dial(“SIP/ken-0982b738”, “SIP/x-lite| 10”) in new stack
[Nov 19 07:56:48] – Called x-lite
[Nov 19 07:56:48] – SIP/x-lite-09830c78 is ringing
[Nov 19 07:56:58] – Nobody picked up in 10000 ms
[Nov 19 07:56:58] == Auto fallthrough, channel ‘SIP/ken-0982b738’ status is 'NOANSWER’
NOTE: Dialed softphone displays “Incoming call on line 1 hung up” and immediately stops ringing…

==================
KEN CALLS PAT SIP DEBUG
(Both Cisco 7970 phones)

*CLI> [Nov 19 08:02:00]
<-- SIP read from 192.168.168.41:49442:
INVITE sip:104@asterisk.jmtg.local SIP/2.0
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bKe92f0c5b
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b256f428f2b28-fdd44109
To: sip:104@asterisk.jmtg.local
Call-ID: 001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41
Max-Forwards: 70
Date: Sun, 19 Nov 2006 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7970G/8.0
Contact: sip:101@192.168.168.41:5060;transport=udp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: “Ken Morley” sip:ken@192.168.168.41;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 277
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 12543 0 IN IP4 192.168.168.41
s=SIP Call
t=0 0
m=audio 23012 RTP/AVP 0 8 18 101
c=IN IP4 192.168.168.41
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

[Nov 19 08:02:00] — (19 headers 13 lines) —
[Nov 19 08:02:00] Using INVITE request as basis request - 001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41
[Nov 19 08:02:00] Sending to 192.168.168.41 : 5060 (non-NAT)
[Nov 19 08:02:00] Reliably Transmitting (no NAT) to 192.168.168.41:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bKe92f0c5b;received=192.168.168.41
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b256f428f2b28-fdd44109
To: sip:104@asterisk.jmtg.local;tag=as1344d053
Call-ID: 001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7b6f181b"
Content-Length: 0


[Nov 19 08:02:00] Scheduling destruction of call ‘001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41’ in 15000 ms
[Nov 19 08:02:00] Found user ‘ken’
[Nov 19 08:02:00]
<-- SIP read from 192.168.168.41:49443:
ACK sip:104@asterisk.jmtg.local SIP/2.0
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bKe92f0c5b
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b256f428f2b28-fdd44109
To: sip:104@asterisk.jmtg.local;tag=as1344d053
Call-ID: 001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41
Date: Sun, 19 Nov 2006 GMT
CSeq: 101 ACK
Content-Length: 0

[Nov 19 08:02:00] — (8 headers 0 lines) —
[Nov 19 08:02:00]
<-- SIP read from 192.168.168.41:49444:
INVITE sip:104@asterisk.jmtg.local SIP/2.0
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bK6ef8e0de
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b256f428f2b28-fdd44109
To: sip:104@asterisk.jmtg.local
Call-ID: 001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41
Max-Forwards: 70
Date: Sun, 19 Nov 2006 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7970G/8.0
Contact: sip:101@192.168.168.41:5060;transport=udp
Proxy-Authorization: Digest username=“ken”,realm=“asterisk”,uri=“sip:104@asterisk.jmtg.local”,response=“779a25ace27ef320fc7c69e2c4042ad0”,nonce=“7b6f181b”,algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: “Ken Morley” sip:ken@192.168.168.41;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 277
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 12543 0 IN IP4 192.168.168.41
s=SIP Call
t=0 0
m=audio 23012 RTP/AVP 0 8 18 101
c=IN IP4 192.168.168.41
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

[Nov 19 08:02:00] — (20 headers 13 lines) —
[Nov 19 08:02:00] Using INVITE request as basis request - 001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41
[Nov 19 08:02:00] Sending to 192.168.168.41 : 5060 (non-NAT)
[Nov 19 08:02:00] Found user ‘ken’
[Nov 19 08:02:00] Found RTP audio format 0
[Nov 19 08:02:00] Found RTP audio format 8
[Nov 19 08:02:00] Found RTP audio format 18
[Nov 19 08:02:00] Found RTP audio format 101
[Nov 19 08:02:00] Peer audio RTP is at port 192.168.168.41:23012
[Nov 19 08:02:00] Found description format PCMU
[Nov 19 08:02:00] Found description format PCMA
[Nov 19 08:02:00] Found description format G729
[Nov 19 08:02:00] Found description format telephone-event
[Nov 19 08:02:00] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Nov 19 08:02:00] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Nov 19 08:02:00] Looking for 104 in default (domain asterisk.jmtg.local)
[Nov 19 08:02:00] list_route: hop: sip:101@192.168.168.41:5060;transport=udp
[Nov 19 08:02:00] Transmitting (no NAT) to 192.168.168.41:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bK6ef8e0de;received=192.168.168.41
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b256f428f2b28-fdd44109
To: sip:104@asterisk.jmtg.local
Call-ID: 001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:104@192.168.168.16
Content-Length: 0


[Nov 19 08:02:00] – Executing Dial(“SIP/ken-0927d3a0”, “SIP/pat| 10”) in new stack
[Nov 19 08:02:00] We’re at 192.168.168.16 port 19244
[Nov 19 08:02:00] Adding codec 0x4 (ulaw) to SDP
[Nov 19 08:02:00] Adding codec 0x2 (gsm) to SDP
[Nov 19 08:02:00] Adding codec 0x8 (alaw) to SDP
[Nov 19 08:02:00] Adding non-codec 0x1 (telephone-event) to SDP
[Nov 19 08:02:00] 13 headers, 12 lines
[Nov 19 08:02:00] Reliably Transmitting (no NAT) to 192.168.168.44:5060:
INVITE sip:104@192.168.168.44:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.168.16:5060;branch=z9hG4bK1e98ad46;rport
From: “Ken Morley” sip:ken@192.168.168.16;tag=as3dcf52c1
To: sip:104@192.168.168.44:5060;transport=udp
Contact: sip:ken@192.168.168.16
Call-ID: 44dcfa335ffd885c0dbfbf420cb0361b@192.168.168.16
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 19 Nov 2006 13:02:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 29746 29746 IN IP4 192.168.168.16
s=session
c=IN IP4 192.168.168.16
t=0 0
m=audio 19244 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


[Nov 19 08:02:00] – Called pat
[Nov 19 08:02:01]
<-- SIP read from 192.168.168.44:49290:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.168.16:5060;branch=z9hG4bK1e98ad46;rport
From: “Ken Morley” sip:ken@192.168.168.16;tag=as3dcf52c1
To: sip:104@192.168.168.44:5060;transport=udp
Call-ID: 44dcfa335ffd885c0dbfbf420cb0361b@192.168.168.16
Date: Sun, 19 Nov 2006 GMT
CSeq: 102 INVITE
Server: Cisco-CP7970G/8.0
Contact: sip:104@192.168.168.44:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: “Pat Morley” sip:pat@192.168.168.44;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0

[Nov 19 08:02:01] — (13 headers 0 lines) —
[Nov 19 08:02:04]
<-- SIP read from 192.168.168.44:49293:
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 192.168.168.16:5060;branch=z9hG4bK17b2cb4a;rport
From: “Ken Morley” sip:ken@192.168.168.16;tag=as17445c18
To: sip:104@192.168.168.44:5060;transport=udp;tag=00164676523725552cf5eab2-aa98ffc5
Call-ID: 57799d3f47139244638fab2b35cf477f@192.168.168.16
Date: Sun, 19 Nov 2006 GMT
CSeq: 102 INVITE
Server: Cisco-CP7970G/8.0
Contact: sip:104@192.168.168.44:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: “Pat Morley” sip:pat@192.168.168.44;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0

[Nov 19 08:02:04] — (13 headers 0 lines) —
[Nov 19 08:02:10] – Nobody picked up in 10000 ms
[Nov 19 08:02:10] Scheduling destruction of call ‘44dcfa335ffd885c0dbfbf420cb0361b@192.168.168.16’ in 32000 ms
[Nov 19 08:02:10] == Auto fallthrough, channel ‘SIP/ken-0927d3a0’ status is ‘NOANSWER’
[Nov 19 08:02:10] Scheduling destruction of call ‘001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41’ in 32000 ms
[Nov 19 08:02:10] Reliably Transmitting (no NAT) to 192.168.168.41:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bK6ef8e0de;received=192.168.168.41
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b256f428f2b28-fdd44109
To: sip:104@asterisk.jmtg.local;tag=as2153ed15
Call-ID: 001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:104@192.168.168.16
Content-Length: 0


[Nov 19 08:02:10]
<-- SIP read from 192.168.168.41:49445:
ACK sip:104@asterisk.jmtg.local SIP/2.0
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bK6ef8e0de
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b256f428f2b28-fdd44109
To: sip:104@asterisk.jmtg.local;tag=as2153ed15
Call-ID: 001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41
Date: Sun, 19 Nov 2006 GMT
CSeq: 102 ACK
Content-Length: 0

[Nov 19 08:02:10] — (8 headers 0 lines) —
[Nov 19 08:02:42] Destroying call ‘44dcfa335ffd885c0dbfbf420cb0361b@192.168.168.16’
[Nov 19 08:02:42] Destroying call ‘001469fb-531b0018-a0c4b652-0c1ffe9f@192.168.168.41’

==================
KEN CALLS X-LITE SIP DEBUG
(Cisco 7970 => X-Lite Softphone)

[Nov 19 08:11:38]
<-- SIP read from 192.168.168.41:49456:
INVITE sip:109@asterisk.jmtg.local SIP/2.0
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bK4482ae32
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b257127594ffd-fdbf7d94
To: sip:109@asterisk.jmtg.local
Call-ID: 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
Max-Forwards: 70
Date: Sun, 19 Nov 2006 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7970G/8.0
Contact: sip:101@192.168.168.41:5060;transport=udp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: “Ken Morley” sip:ken@192.168.168.41;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 277
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 17249 0 IN IP4 192.168.168.41
s=SIP Call
t=0 0
m=audio 19414 RTP/AVP 0 8 18 101
c=IN IP4 192.168.168.41
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

[Nov 19 08:11:38] — (19 headers 13 lines) —
[Nov 19 08:11:38] Using INVITE request as basis request - 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
[Nov 19 08:11:38] Sending to 192.168.168.41 : 5060 (non-NAT)
[Nov 19 08:11:38] Reliably Transmitting (no NAT) to 192.168.168.41:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bK4482ae32;received=192.168.168.41
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b257127594ffd-fdbf7d94
To: sip:109@asterisk.jmtg.local;tag=as6a801914
Call-ID: 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4c3fd056"
Content-Length: 0


[Nov 19 08:11:38] Scheduling destruction of call ‘001469fb-531b001a-46760427-1e4f1f26@192.168.168.41’ in 15000 ms
[Nov 19 08:11:38] Found user ‘ken’
[Nov 19 08:11:38]
<-- SIP read from 192.168.168.41:49457:
ACK sip:109@asterisk.jmtg.local SIP/2.0
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bK4482ae32
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b257127594ffd-fdbf7d94
To: sip:109@asterisk.jmtg.local;tag=as6a801914
Call-ID: 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
Date: Sun, 19 Nov 2006 GMT
CSeq: 101 ACK
Content-Length: 0

[Nov 19 08:11:38] — (8 headers 0 lines) —
[Nov 19 08:11:38]
<-- SIP read from 192.168.168.41:49458:
INVITE sip:109@asterisk.jmtg.local SIP/2.0
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bKda0e6e43
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b257127594ffd-fdbf7d94
To: sip:109@asterisk.jmtg.local
Call-ID: 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
Max-Forwards: 70
Date: Sun, 19 Nov 2006 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7970G/8.0
Contact: sip:101@192.168.168.41:5060;transport=udp
Proxy-Authorization: Digest username=“ken”,realm=“asterisk”,uri=“sip:109@asterisk.jmtg.local”,response=“ae26dfa9824c2b74991c3523423eb8c7”,nonce=“4c3fd056”,algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: “Ken Morley” sip:ken@192.168.168.41;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 277
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 17249 0 IN IP4 192.168.168.41
s=SIP Call
t=0 0
m=audio 19414 RTP/AVP 0 8 18 101
c=IN IP4 192.168.168.41
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

[Nov 19 08:11:38] — (20 headers 13 lines) —
[Nov 19 08:11:38] Using INVITE request as basis request - 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
[Nov 19 08:11:38] Sending to 192.168.168.41 : 5060 (non-NAT)
[Nov 19 08:11:38] Found user ‘ken’
[Nov 19 08:11:38] Found RTP audio format 0
[Nov 19 08:11:38] Found RTP audio format 8
[Nov 19 08:11:38] Found RTP audio format 18
[Nov 19 08:11:38] Found RTP audio format 101
[Nov 19 08:11:38] Peer audio RTP is at port 192.168.168.41:19414
[Nov 19 08:11:38] Found description format PCMU
[Nov 19 08:11:38] Found description format PCMA
[Nov 19 08:11:38] Found description format G729
[Nov 19 08:11:38] Found description format telephone-event
[Nov 19 08:11:38] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Nov 19 08:11:38] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Nov 19 08:11:38] Looking for 109 in default (domain asterisk.jmtg.local)
[Nov 19 08:11:38] list_route: hop: sip:101@192.168.168.41:5060;transport=udp
[Nov 19 08:11:38] Transmitting (no NAT) to 192.168.168.41:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bKda0e6e43;received=192.168.168.41
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b257127594ffd-fdbf7d94
To: sip:109@asterisk.jmtg.local
Call-ID: 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:109@192.168.168.16
Content-Length: 0


[Nov 19 08:11:38] – Executing Dial(“SIP/ken-094a9c98”, “SIP/x-lite| 10”) in new stack
[Nov 19 08:11:38] We’re at 192.168.168.16 port 18716
[Nov 19 08:11:38] Adding codec 0x4 (ulaw) to SDP
[Nov 19 08:11:38] Adding codec 0x2 (gsm) to SDP
[Nov 19 08:11:38] Adding codec 0x8 (alaw) to SDP
[Nov 19 08:11:38] Adding non-codec 0x1 (telephone-event) to SDP
[Nov 19 08:11:38] 13 headers, 12 lines
[Nov 19 08:11:38] Reliably Transmitting (no NAT) to 192.168.168.101:47654:
INVITE sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281 SIP/2.0
Via: SIP/2.0/UDP 192.168.168.16:5060;branch=z9hG4bK4f3811cb;rport
From: “Ken Morley” sip:ken@192.168.168.16;tag=as4f908a14
To: sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281
Contact: sip:ken@192.168.168.16
Call-ID: 07f132312ef12e87480c7cb85487662c@192.168.168.16
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 19 Nov 2006 13:11:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 29776 29776 IN IP4 192.168.168.16
s=session
c=IN IP4 192.168.168.16
t=0 0
m=audio 18716 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


[Nov 19 08:11:38] – Called x-lite
[Nov 19 08:11:38]
<-- SIP read from 192.168.168.101:47654:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.168.16:5060;branch=z9hG4bK4f3811cb;rport=5060
Contact: sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281
To: sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281;tag=c164942e
From: "Ken Morley"sip:ken@192.168.168.16;tag=as4f908a14
Call-ID: 07f132312ef12e87480c7cb85487662c@192.168.168.16
CSeq: 102 INVITE
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0

[Nov 19 08:11:38] — (9 headers 0 lines) —
[Nov 19 08:11:38] – SIP/x-lite-094b25f8 is ringing
[Nov 19 08:11:38] Transmitting (no NAT) to 192.168.168.41:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bKda0e6e43;received=192.168.168.41
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b257127594ffd-fdbf7d94
To: sip:109@asterisk.jmtg.local;tag=as58e3497d
Call-ID: 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:109@192.168.168.16
Content-Length: 0


[Nov 19 08:11:38]
<-- SIP read from 192.168.168.101:47654:

[Nov 19 08:11:38] — (0 headers 1 lines) —
[Nov 19 08:11:48] – Nobody picked up in 10000 ms
[Nov 19 08:11:48] Scheduling destruction of call ‘07f132312ef12e87480c7cb85487662c@192.168.168.16’ in 32000 ms
[Nov 19 08:11:48] Reliably Transmitting (no NAT) to 192.168.168.101:47654:
CANCEL sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281 SIP/2.0
Via: SIP/2.0/UDP 192.168.168.16:5060;branch=z9hG4bK4f3811cb;rport
From: “Ken Morley” sip:ken@192.168.168.16;tag=as4f908a14
To: sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281
Contact: sip:ken@192.168.168.16
Call-ID: 07f132312ef12e87480c7cb85487662c@192.168.168.16
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


[Nov 19 08:11:48] == Auto fallthrough, channel ‘SIP/ken-094a9c98’ status is ‘NOANSWER’
[Nov 19 08:11:48] Scheduling destruction of call ‘001469fb-531b001a-46760427-1e4f1f26@192.168.168.41’ in 32000 ms
[Nov 19 08:11:48] Reliably Transmitting (no NAT) to 192.168.168.41:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bKda0e6e43;received=192.168.168.41
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b257127594ffd-fdbf7d94
To: sip:109@asterisk.jmtg.local;tag=as58e3497d
Call-ID: 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:109@192.168.168.16
Content-Length: 0


[Nov 19 08:11:48]
<-- SIP read from 192.168.168.41:49459:
ACK sip:109@asterisk.jmtg.local SIP/2.0
Via: SIP/2.0/UDP 192.168.168.41:5060;branch=z9hG4bKda0e6e43
From: “Ken Morley” sip:ken@asterisk.jmtg.local;tag=001469fb531b257127594ffd-fdbf7d94
To: sip:109@asterisk.jmtg.local;tag=as58e3497d
Call-ID: 001469fb-531b001a-46760427-1e4f1f26@192.168.168.41
Date: Sun, 19 Nov 2006 GMT
CSeq: 102 ACK
Content-Length: 0

[Nov 19 08:11:48] — (8 headers 0 lines) —
[Nov 19 08:11:48]
<-- SIP read from 192.168.168.101:47654:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.168.16:5060;branch=z9hG4bK4f3811cb;rport=5060
Contact: sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281
To: sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281;tag=c164942e
From: "Ken Morley"sip:ken@192.168.168.16;tag=as4f908a14
Call-ID: 07f132312ef12e87480c7cb85487662c@192.168.168.16
CSeq: 102 CANCEL
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0

[Nov 19 08:11:48] — (9 headers 0 lines) —
[Nov 19 08:11:48]
<-- SIP read from 192.168.168.101:47654:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.168.16:5060;branch=z9hG4bK4f3811cb;rport=5060
To: sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281;tag=c164942e
From: "Ken Morley"sip:ken@192.168.168.16;tag=as4f908a14
Call-ID: 07f132312ef12e87480c7cb85487662c@192.168.168.16
CSeq: 102 INVITE
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 0

[Nov 19 08:11:48] — (8 headers 0 lines) —
[Nov 19 08:11:48] Transmitting (no NAT) to 192.168.168.101:47654:
ACK sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281 SIP/2.0
Via: SIP/2.0/UDP 192.168.168.16:5060;branch=z9hG4bK4f3811cb;rport
From: “Ken Morley” sip:ken@192.168.168.16;tag=as4f908a14
To: sip:x-lite@192.168.168.101:47654;rinstance=d0624052736b3281;tag=c164942e
Contact: sip:ken@192.168.168.16
Call-ID: 07f132312ef12e87480c7cb85487662c@192.168.168.16
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


[Nov 19 08:11:48] Destroying call ‘07f132312ef12e87480c7cb85487662c@192.168.168.16’
[Nov 19 08:12:08]
<-- SIP read from 192.168.168.101:47654:

[Nov 19 08:12:08] — (0 headers 1 lines) —
[Nov 19 08:12:20] Destroying call ‘001469fb-531b001a-46760427-1e4f1f26@192.168.168.41’
[Nov 19 08:12:38]
<-- SIP read from 192.168.168.101:47654: