thr3ads.net - asterisk users - [Asterisk-Users] WAS: Call pickup (*8) on SIP devices. Bug #116 [Oct 2003]

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Bisker, Scott (7805)
2003-Oct-23 14:45 UTC
[Asterisk-Users] WAS: Call pickup (*8) on SIP devices. Bug #116

I've attached two SIP debugs in reference to bug #116.  They are from
today's CVS build. 

1.  pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the
call.  After which, SIP(2) rings for about 30 seconds then stops.

2.  hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before
the call is answered.  

SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom IP500 -- I've also
tried
with SIP(2) being a 7960 as well.

In scenario 2, when SIP(1) hangs up, a CANCEL message is sent to SIP(2).  

In scenario 1, when SIP(3) picks up the call to SIP(2), SIP(2) never
receives a CANCEL message, thus, it continues to ring.  At the end of the
debug, after SIP(2) stop's ringing, it sends 3 Decline messages to the
asterisk PBX.  


If you need any more debug info, let me know.

-sb

 


-------------- next part --------------
*CLI> sip debug
SIP Debugging Enabled
Sip read:
INVITE sip:8719@192.168.1.15;user=ip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
To: <sip:8719@192.168.1.15;user=ip>
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
Date: Thu, 23 Oct 2003 21:23:19 GMT
CSeq: 101 INVITE
User-Agent: CSCO/5
Contact: <sip:5285@192.168.1.84:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 246
Accept: application/sdp
Remote-Party-ID: "5285"
<sip:5285@192.168.1.84>;party=calling;id-type=subscriber;privacy=off;screen=no

v=0
o=Cisco-SIPUA 9972 27311 IN IP4 192.168.1.84
s=SIP Call
c=IN IP4 192.168.1.84
t=0 0
m=audio 31790 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

14 headers, 11 lines
Using latest request as basis request
Sending to 192.168.1.84 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 2147483647, them - 268/0, combined - 268
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
To: <sip:8719@192.168.1.15;user=ip>;tag=as4284ac7e
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk",
nonce="676c94f0"
Content-Length: 0


 to 192.168.1.84:5060
Sip read:
ACK sip:8719@192.168.1.15;user=ip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
To: <sip:8719@192.168.1.15;user=ip>;tag=as4284ac7e
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
Date: Thu, 23 Oct 2003 21:23:19 GMT
CSeq: 101 ACK
Content-Length: 0


8 headers, 0 lines
Sip read:
INVITE sip:8719@192.168.1.15;user=ip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
To: <sip:8719@192.168.1.15;user=ip>
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
Date: Thu, 23 Oct 2003 21:23:19 GMT
CSeq: 102 INVITE
User-Agent: CSCO/5
Contact: <sip:5285@192.168.1.84:5060>
Proxy-Authorization: Digest
username="5285",realm="asterisk",uri="sip:192.168.1.15",response="5025d36a5940ca107c7bdce5aa
1b7e99",nonce="676c94f0",algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 246
Remote-Party-ID: "5285"
<sip:5285@192.168.1.84>;party=calling;id-type=subscriber;privacy=off;screen=no

v=0
o=Cisco-SIPUA 9972 27311 IN IP4 192.168.1.84
s=SIP Call
c=IN IP4 192.168.1.84
t=0 0
m=audio 31790 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

14 headers, 11 lines
Using latest request as basis request
Sending to 192.168.1.84 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 2147483647, them - 268/0, combined - 268
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 8719 in default
list_route: hop: <sip:5285@192.168.1.84:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
To: <sip:8719@192.168.1.15;user=ip>;tag=as710b2362
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8719@192.168.1.15>
Content-Length: 0


 to 192.168.1.84:5060
    -- Executing Macro("SIP/5285-6f79",
"stdexten|8719|SIP/test1") in new stack
    -- Executing DBget("SIP/5285-6f79", "temp=CS/8719") in
new stack
    -- DBget: varname=temp, family=CS, key=8719
    -- DBget: set variable temp to 0
    -- Executing GotoIf("SIP/5285-6f79", "0?s|4") in new
stack
WARNING[229391]: File pbx.c, Line 4442 (pbx_builtin_gotoif): Not taking any
branch
    -- Executing Dial("SIP/5285-6f79", "SIP/test1|20|t") in
new stack
We're at 192.168.1.15 port 11596
Answering with preferred capability 4
Answering with preferred capability 8
Answering with preferred capability 2
Answering with preferred capability 2147483647
11 headers, 9 lines
Reliably Transmitting:
INVITE sip:192.168.1.181 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0f90d445
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as6a2a2007
To: <sip:192.168.1.181>
Contact: <sip:5285@192.168.1.15>
Call-ID: 08cf93d712ecba2703837fed6f933068@192.168.1.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 11901 11901 IN IP4 192.168.1.15
s=session
c=IN IP4 192.168.1.15
t=0 0
m=audio 11596 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
 (no NAT) to 192.168.1.181:5060
    -- Called test1
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0f90d445
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as6a2a2007
To: <sip:192.168.1.181>;tag=C4BAB225-8696114
CSeq: 102 INVITE
Call-ID: 08cf93d712ecba2703837fed6f933068@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0


9 headers, 0 lines
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0f90d445
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as6a2a2007
To: <sip:192.168.1.181>;tag=C4BAB225-8696114
CSeq: 102 INVITE
Call-ID: 08cf93d712ecba2703837fed6f933068@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0


9 headers, 0 lines
    -- SIP/test1-957c is ringing
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
To: <sip:8719@192.168.1.15;user=ip>;tag=as710b2362
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8719@192.168.1.15>
Content-Length: 0


 to 192.168.1.84:5060
Sip read:
INVITE sip:*8@192.168.1.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060
From: "5244"
<sip:5244@192.168.1.15>;tag=000dbc260f52000e4ec50e12-2b5f3c98
To: <sip:*8@192.168.1.15>
Call-ID: 000dbc26-0f52000f-18618b68-3bced9d6@192.168.1.165
Date: Thu, 23 Oct 2003 21:23:22 GMT
CSeq: 101 INVITE
User-Agent: CSCO/5
Contact: <sip:5244@192.168.1.165:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 248
Accept: application/sdp
Remote-Party-ID: "5244"
<sip:5244@192.168.1.165>;party=calling;id-type=subscriber;privacy=off;screen=no

v=0
o=Cisco-SIPUA 4177 23999 IN IP4 192.168.1.165
s=SIP Call
c=IN IP4 192.168.1.165
t=0 0
m=audio 27440 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

14 headers, 11 lines
Using latest request as basis request
Sending to 192.168.1.165 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 2147483647, them - 268/0, combined - 268
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.165:5060
From: "5244"
<sip:5244@192.168.1.15>;tag=000dbc260f52000e4ec50e12-2b5f3c98
To: <sip:*8@192.168.1.15>;tag=as7095d2fe
Call-ID: 000dbc26-0f52000f-18618b68-3bced9d6@192.168.1.165
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk",
nonce="1a02dcc9"
Content-Length: 0


 to 192.168.1.165:5060
Sip read:
ACK sip:*8@192.168.1.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060
From: "5244"
<sip:5244@192.168.1.15>;tag=000dbc260f52000e4ec50e12-2b5f3c98
To: <sip:*8@192.168.1.15>;tag=as7095d2fe
Call-ID: 000dbc26-0f52000f-18618b68-3bced9d6@192.168.1.165
Date: Thu, 23 Oct 2003 21:23:22 GMT
CSeq: 101 ACK
Content-Length: 0


8 headers, 0 lines
Sip read:
INVITE sip:*8@192.168.1.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060
From: "5244"
<sip:5244@192.168.1.15>;tag=000dbc260f52000e4ec50e12-2b5f3c98
To: <sip:*8@192.168.1.15>
Call-ID: 000dbc26-0f52000f-18618b68-3bced9d6@192.168.1.165
Date: Thu, 23 Oct 2003 21:23:22 GMT
CSeq: 102 INVITE
User-Agent: CSCO/5
Contact: <sip:5244@192.168.1.165:5060>
Proxy-Authorization: Digest
username="5244",realm="asterisk",uri="sip:192.168.1.15",response="8fa65beae5dce87747e42f32b8
0a88d7",nonce="1a02dcc9",algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 248
Remote-Party-ID: "5244"
<sip:5244@192.168.1.165>;party=calling;id-type=subscriber;privacy=off;screen=no

v=0
o=Cisco-SIPUA 4177 23999 IN IP4 192.168.1.165
s=SIP Call
c=IN IP4 192.168.1.165
t=0 0
m=audio 27440 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

14 headers, 11 lines
Using latest request as basis request
Sending to 192.168.1.165 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 2147483647, them - 268/0, combined - 268
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for *8 in default
list_route: hop: <sip:5244@192.168.1.165:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.165:5060
From: "5244"
<sip:5244@192.168.1.15>;tag=000dbc260f52000e4ec50e12-2b5f3c98
To: <sip:*8@192.168.1.15>;tag=as54d9a268
Call-ID: 000dbc26-0f52000f-18618b68-3bced9d6@192.168.1.165
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*8@192.168.1.15>
Content-Length: 0


 to 192.168.1.165:5060
We're at 192.168.1.15 port 10864
Answering with preferred capability 4
Answering with preferred capability 8
Answering with preferred capability 2
Answering with preferred capability 2147483647
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.165:5060
From: "5244"
<sip:5244@192.168.1.15>;tag=000dbc260f52000e4ec50e12-2b5f3c98
To: <sip:*8@192.168.1.15>;tag=as54d9a268
Call-ID: 000dbc26-0f52000f-18618b68-3bced9d6@192.168.1.165
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*8@192.168.1.15>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 11893 11893 IN IP4 192.168.1.15
s=session
c=IN IP4 192.168.1.15
t=0 0
m=audio 10864 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000

 to 192.168.1.165:5060
    -- SIP/5244-800f answered SIP/5285-6f79
We're at 192.168.1.15 port 12018
Answering with preferred capability 4
Answering with preferred capability 8
Answering with preferred capability 2
Answering with preferred capability 2147483647
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
To: <sip:8719@192.168.1.15;user=ip>;tag=as710b2362
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8719@192.168.1.15>
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 11901 11901 IN IP4 192.168.1.15
s=session
c=IN IP4 192.168.1.15
t=0 0
m=audio 12018 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to 192.168.1.84:5060
    -- Attempting native bridge of SIP/5285-6f79 and SIP/5244-800f
Sip read:
ACK sip:8719@192.168.1.15:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
To: <sip:8719@192.168.1.15;user=ip>;tag=as710b2362
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
Date: Thu, 23 Oct 2003 21:23:22 GMT
CSeq: 102 ACK
User-Agent: CSCO/5
Content-Length: 0


9 headers, 0 lines
Sip read:
ACK sip:*8@192.168.1.15:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060
From: "5244"
<sip:5244@192.168.1.15>;tag=000dbc260f52000e4ec50e12-2b5f3c98
To: <sip:*8@192.168.1.15>;tag=as54d9a268
Call-ID: 000dbc26-0f52000f-18618b68-3bced9d6@192.168.1.165
Date: Thu, 23 Oct 2003 21:23:22 GMT
CSeq: 102 ACK
User-Agent: CSCO/5
Content-Length: 0


9 headers, 0 lines
Sip read:
BYE sip:*8@192.168.1.15:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.165:5060
From: "5244"
<sip:5244@192.168.1.15>;tag=000dbc260f52000e4ec50e12-2b5f3c98
To: <sip:*8@192.168.1.15>;tag=as54d9a268
Call-ID: 000dbc26-0f52000f-18618b68-3bced9d6@192.168.1.165
Date: Thu, 23 Oct 2003 21:23:25 GMT
CSeq: 103 BYE
User-Agent: CSCO/5
Content-Length: 0
Proxy-Authorization: Digest
username="5244",realm="asterisk",uri="sip:192.168.1.15",response="2919a5fe59b221ab28d194dc9f
7707bb",nonce="1a02dcc9",algorithm=md5


10 headers, 0 lines
Sending to 192.168.1.165 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.165:5060
From: "5244"
<sip:5244@192.168.1.15>;tag=000dbc260f52000e4ec50e12-2b5f3c98
To: <sip:*8@192.168.1.15>;tag=as54d9a268
Call-ID: 000dbc26-0f52000f-18618b68-3bced9d6@192.168.1.165
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*8@192.168.1.15>
Content-Length: 0


 to 192.168.1.165:5060
  == Spawn extension (macro-stdexten, s, 3) exited non-zero on
'SIP/5285-6f79' in macro 'stdexten'
  == Spawn extension (default, 8719, 1) exited non-zero on
'SIP/5285-6f79'
set_destination: Parsing <sip:5285@192.168.1.84:5060> for address/port to
send to
set_destination: set destination to 192.168.1.84, port 5060
Reliably Transmitting:
BYE sip:5285@192.168.1.84:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK1519e066
From: <sip:8719@192.168.1.15;user=ip>;tag=as710b2362
To: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
Contact: <sip:8719@192.168.1.15>
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.1.84:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK1519e066
From: <sip:8719@192.168.1.15;user=ip>;tag=as710b2362
To: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55
Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84
Date: Thu, 23 Oct 2003 21:23:25 GMT
CSeq: 102 BYE
Server: CSCO/5
Content-Length: 0


9 headers, 0 lines
Message is BYE

Sip read:
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0fbc5360
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as3bfef546
To: <sip:192.168.1.181>;tag=735876E9-69B238D8
CSeq: 102 INVITE
Call-ID: 10fbe2d93fe136f052bf474339987f96@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0


9 headers, 0 lines
Sip read:
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0fbc5360
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as3bfef546
To: <sip:192.168.1.181>;tag=735876E9-69B238D8
CSeq: 102 INVITE
Call-ID: 10fbe2d93fe136f052bf474339987f96@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0


9 headers, 0 lines
Sip read:
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0fbc5360
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as3bfef546
To: <sip:192.168.1.181>;tag=735876E9-69B238D8
CSeq: 102 INVITE
Call-ID: 10fbe2d93fe136f052bf474339987f96@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0


9 headers, 0 lines
Sip read:
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0fbc5360
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as3bfef546
To: <sip:192.168.1.181>;tag=735876E9-69B238D8
CSeq: 102 INVITE
Call-ID: 10fbe2d93fe136f052bf474339987f96@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0

*CLI> 
*CLI>
-------------- next part --------------
Sip read:
INVITE sip:8719@192.168.1.15;user=ip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
Date: Thu, 23 Oct 2003 21:32:07 GMT
CSeq: 101 INVITE
User-Agent: CSCO/5
Contact: <sip:5285@192.168.1.84:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 245
Accept: application/sdp
Remote-Party-ID: "5285"
<sip:5285@192.168.1.84>;party=calling;id-type=subscriber;privacy=off;screen=no

v=0
o=Cisco-SIPUA 4398 5331 IN IP4 192.168.1.84
s=SIP Call
c=IN IP4 192.168.1.84
t=0 0
m=audio 31794 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

14 headers, 11 lines
Using latest request as basis request
Sending to 192.168.1.84 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 2147483647, them - 268/0, combined - 268
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>;tag=as44c70ca5
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk",
nonce="32c397e0"
Content-Length: 0


 to 192.168.1.84:5060
Sip read:
ACK sip:8719@192.168.1.15;user=ip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>;tag=as44c70ca5
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
Date: Thu, 23 Oct 2003 21:32:07 GMT
CSeq: 101 ACK
Content-Length: 0


8 headers, 0 lines
Sip read:
INVITE sip:8719@192.168.1.15;user=ip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
Date: Thu, 23 Oct 2003 21:32:08 GMT
CSeq: 102 INVITE
User-Agent: CSCO/5
Contact: <sip:5285@192.168.1.84:5060>
Proxy-Authorization: Digest
username="5285",realm="asterisk",uri="sip:192.168.1.15",response="592feb69cf8aa0edc860d606
e31bda",nonce="32c397e0",algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 245
Remote-Party-ID: "5285"
<sip:5285@192.168.1.84>;party=calling;id-type=subscriber;privacy=off;screen=no

v=0
o=Cisco-SIPUA 4398 5331 IN IP4 192.168.1.84
s=SIP Call
c=IN IP4 192.168.1.84
t=0 0
m=audio 31794 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

14 headers, 11 lines
Using latest request as basis request
Sending to 192.168.1.84 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 2147483647, them - 268/0, combined - 268
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 8719 in default
list_route: hop: <sip:5285@192.168.1.84:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>;tag=as3f9c6d42
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8719@192.168.1.15>
Content-Length: 0


 to 192.168.1.84:5060
    -- Executing Macro("SIP/5285-4fb9",
"stdexten|8719|SIP/test1") in new stack
    -- Executing DBget("SIP/5285-4fb9", "temp=CS/8719") in
new stack
    -- DBget: varname=temp, family=CS, key=8719
    -- DBget: set variable temp to 0
    -- Executing GotoIf("SIP/5285-4fb9", "0?s|4") in new
stack
WARNING[278543]: File pbx.c, Line 4442 (pbx_builtin_gotoif): Not taking any
branch
    -- Executing Dial("SIP/5285-4fb9", "SIP/test1|20|t") in
new stack
We're at 192.168.1.15 port 18862
Answering with preferred capability 4
Answering with preferred capability 8
Answering with preferred capability 2
Answering with preferred capability 2147483647
11 headers, 9 lines
Reliably Transmitting:
INVITE sip:192.168.1.181 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK4d374d61
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as21e2c702
To: <sip:192.168.1.181>
Contact: <sip:5285@192.168.1.15>
Call-ID: 15ddf8eb6505dc3f400bbbe171e88e82@192.168.1.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 11904 11904 IN IP4 192.168.1.15
s=session
c=IN IP4 192.168.1.15
t=0 0
m=audio 18862 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
 (no NAT) to 192.168.1.181:5060
    -- Called test1
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK4d374d61
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as21e2c702
To: <sip:192.168.1.181>;tag=C0D86FAD-7D04449C
CSeq: 102 INVITE
Call-ID: 15ddf8eb6505dc3f400bbbe171e88e82@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0


9 headers, 0 lines
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK4d374d61
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as21e2c702
To: <sip:192.168.1.181>;tag=C0D86FAD-7D04449C
CSeq: 102 INVITE
Call-ID: 15ddf8eb6505dc3f400bbbe171e88e82@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0


9 headers, 0 lines
    -- SIP/test1-7f77 is ringing
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>;tag=as3f9c6d42
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8719@192.168.1.15>
Content-Length: 0


 to 192.168.1.84:5060
Sip read:
CANCEL sip:8719@192.168.1.15;user=ip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
Date: Thu, 23 Oct 2003 21:32:13 GMT
CSeq: 102 CANCEL
User-Agent: CSCO/5
Content-Length: 0
Proxy-Authorization: Digest
username="5285",realm="asterisk",uri="sip:192.168.1.15",response="233df3d9ad345418ac9e1d4a
ad8598",nonce="32c397e0",algorithm=md5


10 headers, 0 lines
Sending to 192.168.1.84 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>;tag=as3f9c6d42
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8719@192.168.1.15>
Content-Length: 0


 to 192.168.1.84:5060
Reliably Transmitting (no NAT):
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>;tag=as3f9c6d42
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8719@192.168.1.15>
Content-Length: 0


 to 192.168.1.84:5060
Reliably Transmitting:
CANCEL sip:192.168.1.181 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK4d374d61
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as21e2c702
To: <sip:192.168.1.181>
Contact: <sip:5285@192.168.1.15>
Call-ID: 15ddf8eb6505dc3f400bbbe171e88e82@192.168.1.15
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.1.181:5060
  == Spawn extension (macro-stdexten, s, 3) exited non-zero on
'SIP/5285-4fb9' in macro 'stdexten'
  == Spawn extension (default, s, 1) exited non-zero on 'SIP/5285-4fb9'
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK4d374d61
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as21e2c702
To: <sip:192.168.1.181>
CSeq: 102 CANCEL
Call-ID: 15ddf8eb6505dc3f400bbbe171e88e82@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0


9 headers, 0 lines
Sip read:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK4d374d61
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as21e2c702
To: <sip:192.168.1.181>;tag=C0D86FAD-7D04449C
CSeq: 102 INVITE
Call-ID: 15ddf8eb6505dc3f400bbbe171e88e82@192.168.1.15
Contact:<sip:192.168.1.181>
User-Agent: PolycomSoundPointIP-UA/1.0.4
Content-Length: 0


9 headers, 0 lines
Transmitting:
ACK sip:192.168.1.181 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK4d374d61
From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as21e2c702
To: <sip:192.168.1.181>;tag=C0D86FAD-7D04449C
Contact: <sip:5285@192.168.1.15>
Call-ID: 15ddf8eb6505dc3f400bbbe171e88e82@192.168.1.15
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.1.181:5060
Sip read:
ACK sip:8719@192.168.1.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5060
From: "5285"
<sip:5285@192.168.1.15>;tag=000d287e269a00114a851a42-39e47ff2
To: <sip:8719@192.168.1.15;user=ip>;tag=as3f9c6d42
Call-ID: 000d287e-269a0016-6f20d1fb-534a70b0@192.168.1.84
Date: Thu, 23 Oct 2003 21:32:13 GMT
CSeq: 102 ACK
Content-Length: 0


8 headers, 0 lines

*CLI>
Possibly Parallel Threads

Search for more possibly parallel threads
asterisk users - Oct 2003 - WAS: Call pickup (*8) on SIP devices. Bug #116

[Asterisk-Users] WAS: Call pickup (*8) on SIP devices. Bug #116

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Wisdom of the Ancients
