Hello.
I have a strange problem. Its not possible to pickup a call that was placed
with a Siemens SL75 Wlan. When this phone calls an internal number and i try
to pickup (*8) the call from my phone i get nothing. It seems i have the call
for one second or so but after that the call is being cancelled. No problems
with other phones (polycom, grandstream). Attached the complete sip debug log
of such a call. Any help would be high appreciated.
regards
t.
asterix*CLI> sip debug
SIP Debugging enabled
asterix*CLI>
<-- SIP read from 217.10.79.9:5060:
--- (0 headers 0 lines) Nat keepalive ---
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
INVITE sip:119 at 192.168.150.151 SIP/2.0
Max-Forwards: 70
Content-Length: 293
Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK6abcc4ace
Call-ID: 94cba353ee1163b
From: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
To: sip:119 at 192.168.150.151
CSeq: 2078851383 INVITE
Supported: timer
Session-Expires: 7200
Allow-Events: talk, hold, conference
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO
Content-Type: application/sdp
Contact: Steffen <sip:116 at 192.168.150.51:5060;transport=udp>
Supported: replaces
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
v=0
o=MxSIP 0 1730916047 IN IP4 192.168.150.51
s=SIP Call
c=IN IP4 192.168.150.51
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (16 headers 13 lines) ---
Using INVITE request as basis request - 94cba353ee1163b
Sending to 192.168.150.51 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.150.51:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.150.51:5060;branch=z9hG4bK6abcc4ace;received=192.168.150.51
From: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
To: sip:119 at 192.168.150.151;tag=as06838deb
Call-ID: 94cba353ee1163b
CSeq: 2078851383 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="7e84319d"
Content-Length: 0
---
Scheduling destruction of call '94cba353ee1163b' in 15000 ms
Found user '116'
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
ACK sip:119 at 192.168.150.151 SIP/2.0
Max-Forwards: 70
Content-Length: 0
Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK6abcc4ace
Call-ID: 94cba353ee1163b
From: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
To: sip:119 at 192.168.150.151;tag=as06838deb
CSeq: 2078851383 ACK
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
--- (9 headers 0 lines) ---
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
INVITE sip:119 at 192.168.150.151 SIP/2.0
Max-Forwards: 70
Content-Length: 293
Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK2e49558fa
Call-ID: 94cba353ee1163b
From: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
To: sip:119 at 192.168.150.151
CSeq: 2078851384 INVITE
Supported: timer
Session-Expires: 7200
Allow-Events: talk, hold, conference
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO
Content-Type: application/sdp
Proxy-Authorization:Digest
response="ea33e742f1b16d49344c67d8cc980a16",username="116",realm="asterisk",nonce="7e84319d",algorithm=MD5,uri="sip:119
at 192.168.150.151"
Supported: replaces
Contact: Steffen <sip:116 at 192.168.150.51:5060;transport=udp>
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
v=0
o=MxSIP 0 1730916047 IN IP4 192.168.150.51
s=SIP Call
c=IN IP4 192.168.150.51
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (17 headers 13 lines) ---
Using INVITE request as basis request - 94cba353ee1163b
Sending to 192.168.150.51 : 5060 (non-NAT)
Found user '116'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Peer audio RTP is at port 192.168.150.51:5004
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G722
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|
g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 119 in default (domain 192.168.150.151)
list_route: hop: <sip:116 at 192.168.150.51:5060;transport=udp>
Transmitting (no NAT) to 192.168.150.51:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.150.51:5060;branch=z9hG4bK2e49558fa;received=192.168.150.51
From: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
To: sip:119 at 192.168.150.151
Call-ID: 94cba353ee1163b
CSeq: 2078851384 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:119 at 192.168.150.151>
Content-Length: 0
---
We're at 192.168.150.151 port 16216
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.150.51:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.150.51:5060;branch=z9hG4bK2e49558fa;received=192.168.150.51
From: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
To: sip:119 at 192.168.150.151;tag=as36ac8450
Call-ID: 94cba353ee1163b
CSeq: 2078851384 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:119 at 192.168.150.151>
Content-Type: application/sdp
Content-Length: 246
v=0
o=root 24922 24922 IN IP4 192.168.150.151
s=session
c=IN IP4 192.168.150.151
t=0 0
m=audio 16216 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
We're at 192.168.150.151 port 7280
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 192.168.150.11:5060:
INVITE sip:119 at 192.168.150.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport
From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6
To: <sip:119 at 192.168.150.11:5060>
Contact: <sip:116 at 192.168.150.151>
Call-ID: 17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 20 Nov 2007 09:41:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 245
v=0
o=root 24922 24922 IN IP4 192.168.150.151
s=session
c=IN IP4 192.168.150.151
t=0 0
m=audio 7280 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
asterix*CLI>
<-- SIP read from 192.168.150.11:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport
From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6
To: <sip:119 at 192.168.150.11:5060>
Call-ID: 17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.1.1.14
Content-Length: 0
--- (8 headers 0 lines) ---
asterix*CLI>
<-- SIP read from 192.168.150.11:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport
From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6
To: <sip:119 at 192.168.150.11:5060>;tag=1fc925a8554a4f7a
Call-ID: 17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.1.1.14
Contact: <sip:119 at 192.168.150.11:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
--- (10 headers 0 lines) ---
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
ACK sip:119 at 192.168.150.151 SIP/2.0
Max-Forwards: 70
Content-Length: 0
Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK40bfb60e5
Call-ID: 94cba353ee1163b
From: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
To: sip:119 at 192.168.150.151;tag=as36ac8450
CSeq: 2078851384 ACK
Proxy-Authorization:Digest
response="ea33e742f1b16d49344c67d8cc980a16",username="116",realm="asterisk",nonce="7e84319d",algorithm=MD5,uri="sip:119
at 192.168.150.151"
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
--- (10 headers 0 lines) ---
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
--- (0 headers 0 lines) Nat keepalive ---
Destroying call '0e8f3e0a3882447978561e5122ffcb02 at 192.168.150.151'
asterix*CLI>
<-- SIP read from 192.168.150.43:5060:
INVITE sip:*8 at 192.168.150.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bK4c66703c19E1401F
From: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
To: <sip:*8 at 192.168.150.151>
CSeq: 1 INVITE
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
Contact: <sip:118 at 192.168.150.43>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 1195480065 1195480065 IN IP4 192.168.150.43
s=Polycom IP Phone
c=IN IP4 192.168.150.43
t=0 0
m=audio 2222 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 11 lines) ---
Using INVITE request as basis request -
fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
Sending to 192.168.150.43 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.150.43:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.150.43;branch=z9hG4bK4c66703c19E1401F;received=192.168.150.43
From: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
To: <sip:*8 at 192.168.150.151>;tag=as41909f67
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="059c6efc"
Content-Length: 0
---
Scheduling destruction of call 'fac1920a-b3f038bd-ac4978f8 at
192.168.150.43' in
15000 ms
Found user '118'
asterix*CLI>
<-- SIP read from 192.168.150.43:5060:
ACK sip:*8 at 192.168.150.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bK4c66703c19E1401F
From: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
To: <sip:*8 at 192.168.150.151>;tag=as41909f67
CSeq: 1 ACK
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
Contact: <sip:118 at 192.168.150.43>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines) ---
asterix*CLI>
<-- SIP read from 192.168.150.43:5060:
INVITE sip:*8 at 192.168.150.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bKf571a0598596EE34
From: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
To: <sip:*8 at 192.168.150.151>
CSeq: 2 INVITE
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
Contact: <sip:118 at 192.168.150.43>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="118",
realm="asterisk",
nonce="059c6efc", uri="sip:*8 at 192.168.150.151:5060",
response="3025cdb052cbf156933a61f6470dd21d", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 1195480065 1195480065 IN IP4 192.168.150.43
s=Polycom IP Phone
c=IN IP4 192.168.150.43
t=0 0
m=audio 2222 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
--- (15 headers 11 lines) ---
Using INVITE request as basis request -
fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
Sending to 192.168.150.43 : 5060 (non-NAT)
Found user '118'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.150.43:2222
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|
g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for *8 in default (domain 192.168.150.151)
list_route: hop: <sip:118 at 192.168.150.43>
Transmitting (no NAT) to 192.168.150.43:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.150.43;branch=z9hG4bKf571a0598596EE34;received=192.168.150.43
From: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
To: <sip:*8 at 192.168.150.151>
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*8 at 192.168.150.151>
Content-Length: 0
---
We're at 192.168.150.151 port 13978
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.150.43:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.150.43;branch=z9hG4bKf571a0598596EE34;received=192.168.150.43
From: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
To: <sip:*8 at 192.168.150.151>;tag=as45b8e08e
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*8 at 192.168.150.151>
Content-Type: application/sdp
Content-Length: 246
v=0
o=root 24922 24922 IN IP4 192.168.150.151
s=session
c=IN IP4 192.168.150.151
t=0 0
m=audio 13978 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Scheduling destruction of
call '17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151' in 32000 ms
Reliably Transmitting (no NAT) to 192.168.150.11:5060:
CANCEL sip:119 at 192.168.150.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport
From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6
To: <sip:119 at 192.168.150.11:5060>
Call-ID: 17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
set_destination: Parsing <sip:116 at 192.168.150.51:5060;transport=udp>
for
address/port to send to
set_destination: set destination to 192.168.150.51, port 5060
We're at 192.168.150.151 port 16216
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to 192.168.150.51:5060:
INVITE sip:116 at 192.168.150.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport
From: sip:119 at 192.168.150.151;tag=as36ac8450
To: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
Contact: <sip:119 at 192.168.150.151>
Call-ID: 94cba353ee1163b
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 24922 24923 IN IP4 192.168.150.43
s=session
c=IN IP4 192.168.150.43
t=0 0
m=audio 2222 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
asterix*CLI>
<-- SIP read from 192.168.150.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport
From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6
To: <sip:119 at 192.168.150.11:5060>;tag=1fc925a8554a4f7a
Call-ID: 17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151
CSeq: 102 CANCEL
User-Agent: Grandstream BT200 1.1.1.14
Contact: <sip:119 at 192.168.150.11:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0
--- (11 headers 0 lines) ---
asterix*CLI>
<-- SIP read from 192.168.150.11:5060:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport
From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6
To: <sip:119 at 192.168.150.11:5060>;tag=1fc925a8554a4f7a
Call-ID: 17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.1.1.14
Content-Length: 0
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.150.11:5060:
ACK sip:119 at 192.168.150.11:5060 SIP/2.0
ia: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport
From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6
To: <sip:119 at 192.168.150.11:5060>;tag=1fc925a8554a4f7a
Contact: <sip:116 at 192.168.150.151>
Call-ID: 17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Destroying call '17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151'
asterix*CLI>
<-- SIP read from 192.168.150.43:5060:
ACK sip:*8 at 192.168.150.151 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bKdccbff09857C5B3
From: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
To: <sip:*8 at 192.168.150.151>;tag=as45b8e08e
CSeq: 2 ACK
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
Contact: <sip:118 at 192.168.150.43>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037
Proxy-Authorization: Digest username="118",
realm="asterisk",
nonce="059c6efc", uri="sip:*8 at 192.168.150.151:5060",
response="3025cdb052cbf156933a61f6470dd21d", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:118 at 192.168.150.43> for address/port to
send to
set_destination: set destination to 192.168.150.43, port 5060
We're at 192.168.150.151 port 13978
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 15 lines
Reliably Transmitting (no NAT) to 192.168.150.43:5060:
INVITE sip:118 at 192.168.150.43 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK7055cc60;rport
From: <sip:*8 at 192.168.150.151>;tag=as45b8e08e
To: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
Contact: <sip:*8 at 192.168.150.151>
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 24922 24923 IN IP4 192.168.150.51
s=session
c=IN IP4 192.168.150.51
t=0 0
m=audio 5004 RTP/AVP 8 0 4 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
asterix*CLI>
<-- SIP read from 192.168.150.43:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK7055cc60;rport
From: <sip:*8 at 192.168.150.151>;tag=as45b8e08e
To: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
CSeq: 102 INVITE
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
Contact: <sip:118 at 192.168.150.43>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037
Content-Type: application/sdp
Content-Length: 191
v=0
o=- 1195480065 1195480066 IN IP4 192.168.150.43
s=Polycom IP Phone
c=IN IP4 192.168.150.43
t=0 0
m=audio 2222 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
--- (11 headers 8 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.150.43:2222
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0
(nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:118 at 192.168.150.43>
set_destination: Parsing <sip:118 at 192.168.150.43> for address/port to
send to
set_destination: set destination to 192.168.150.43, port 5060
Transmitting (no NAT) to 192.168.150.43:5060:
ACK sip:118 at 192.168.150.43 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK55ee2bfa;rport
From: <sip:*8 at 192.168.150.151>;tag=as45b8e08e
To: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
Contact: <sip:*8 at 192.168.150.151>
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
SIP/2.0 100 Trying
Call-ID: 94cba353ee1163b
CSeq: 102 INVITE
From: sip:119 at 192.168.150.151;tag=as36ac8450
To: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport
Content-Length: 0
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:116 at 192.168.150.51:5060;transport=udp>
for
address/port to send to
set_destination: set destination to 192.168.150.51, port 5060
We're at 192.168.150.151 port 16216
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.150.51:5060:
INVITE sip:116 at 192.168.150.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK53eb19ca;rport
From: sip:119 at 192.168.150.151;tag=as36ac8450
To: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
Contact: <sip:119 at 192.168.150.151>
Call-ID: 94cba353ee1163b
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 219
v=0
o=root 24922 24924 IN IP4 192.168.150.43
s=session
c=IN IP4 192.168.150.43
t=0 0
m=audio 2222 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
SIP/2.0 500 Server Internal Error
Call-ID: 94cba353ee1163b
CSeq: 103 INVITE
From: sip:119 at 192.168.150.151;tag=as36ac8450
To: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK53eb19ca;rport
Content-Length: 0
Retry-After: 8
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:116 at 192.168.150.51:5060;transport=udp>
for
address/port to send to
set_destination: set destination to 192.168.150.51, port 5060
Transmitting (no NAT) to 192.168.150.51:5060:
ACK sip:116 at 192.168.150.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK53eb19ca;rport
From: sip:119 at 192.168.150.151;tag=as36ac8450
To: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
Contact: <sip:119 at 192.168.150.151>
Call-ID: 94cba353ee1163b
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of call 'fac1920a-b3f038bd-ac4978f8 at
192.168.150.43' in
32000 ms
set_destination: Parsing <sip:118 at 192.168.150.43> for address/port to
send to
set_destination: set destination to 192.168.150.43, port 5060
Reliably Transmitting (no NAT) to 192.168.150.43:5060:
BYE sip:118 at 192.168.150.43 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK3b752b3b;rport
From: <sip:*8 at 192.168.150.151>;tag=as45b8e08e
To: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
asterix*CLI>
<-- SIP read from 192.168.150.43:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK3b752b3b;rport
From: <sip:*8 at 192.168.150.151>;tag=as45b8e08e
To: "Thomas Stein" <sip:118 at
192.168.150.151>;tag=ED14397B-9031B306
CSeq: 103 BYE
Call-ID: fac1920a-b3f038bd-ac4978f8 at 192.168.150.43
Contact: <sip:118 at 192.168.150.43>
User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037
Content-Length: 0
--- (9 headers 0 lines) ---
Destroying call 'fac1920a-b3f038bd-ac4978f8 at 192.168.150.43'
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
SIP/2.0 200 OK
Call-ID: 94cba353ee1163b
CSeq: 102 INVITE
From: sip:119 at 192.168.150.151;tag=as36ac8450
To: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport
Content-Length: 257
Session-Expires: 7200;refresher=uas
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO
Content-Type: application/sdp
Supported: replaces
Contact: Steffen <sip:116 at 192.168.150.51:5060;transport=udp>
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
v=0
o=MxSIP 0 1730916047 IN IP4 192.168.150.51
s=SIP Call
c=IN IP4 192.168.150.51
t=0 0
m=audio 5004 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (13 headers 12 lines) ---
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
SIP/2.0 200 OK
Call-ID: 94cba353ee1163b
CSeq: 102 INVITE
From: sip:119 at 192.168.150.151;tag=as36ac8450
To: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport
Content-Length: 257
Session-Expires: 7200;refresher=uas
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO
Content-Type: application/sdp
Supported: replaces
Contact: Steffen <sip:116 at 192.168.150.51:5060;transport=udp>
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
v=0
o=MxSIP 0 1730916047 IN IP4 192.168.150.51
s=SIP Call
c=IN IP4 192.168.150.51
t=0 0
m=audio 5004 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (13 headers 12 lines) ---
asterix*CLI>
<-- SIP read from 192.168.150.51:5060:
SIP/2.0 200 OK
Call-ID: 94cba353ee1163b
CSeq: 102 INVITE
From: sip:119 at 192.168.150.151;tag=as36ac8450
To: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport
Content-Length: 257
Session-Expires: 7200;refresher=uas
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO
Content-Type: application/sdp
Supported: replaces
Contact: Steffen <sip:116 at 192.168.150.51:5060;transport=udp>
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
v=0
o=MxSIP 0 1730916047 IN IP4 192.168.150.51
s=SIP Call
c=IN IP4 192.168.150.51
t=0 0
m=audio 5004 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (13 headers 12 lines) ---
asterix*CLI>
<-- SIP read from 217.10.79.9:5060:
--- (0 headers 0 lines) Nat keepalive ---
asterix*CLI> sip de
<-- SIP read from 192.168.150.51:5060:
SIP/2.0 200 OK
Call-ID: 94cba353ee1163b
CSeq: 102 INVITE
From: sip:119 at 192.168.150.151;tag=as36ac8450
To: Steffen <sip:116 at 192.168.150.151>;tag=19a39a2dc7a54cd
Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport
Content-Length: 257
Session-Expires: 7200;refresher=uas
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO
Content-Type: application/sdp
Supported: replaces
Contact: Steffen <sip:116 at 192.168.150.51:5060;transport=udp>
User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8
v=0
o=MxSIP 0 1730916047 IN IP4 192.168.150.51
s=SIP Call
c=IN IP4 192.168.150.51
t=0 0
m=audio 5004 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (13 headers 12 lines) ---
asterix*CLI> sip no debug
SIP Debugging Disabled
--
knowledgeTools? ... managing complexity.
--------------------------------------------------
knowledgeTools International GmbH
Wallstra?e 15 / 15 a
10179 Berlin
Fon: +49 30 726 169 20
Fax: +49 30 726 169 249
thomas.stein at knowledgetools.de
www.knowledgetools.de
Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378
Gesch?ftsf?hrer: Oliver Seyboldt, Reinhard Kunz
--------------------------------------------------
This eMail communication (and any attachment/s) may contain confidential or
privileged information and is intended only for the individual(s) or entity
named above and to others who have been specifically authorized to receive
it. If you are not the intended recipient, please do not read, copy, use or
disclose the contents of this communication to others. Please notify the
sender that you have received this e-mail in error by reply e-mail, and
delete the e-mail subsequently. Thank you.
