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gkservice

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Hallo,

wir haben einen Kirk 600V3 und einen Asterisk im _selben_ Netz trotzdem passiert folgendes:
2 Dect Phones rufen sich gegenseitig an und A hört B nicht wobei B trotzdem A hören kann
Code: 
Transmitting (no NAT) to 192.168.200.25:16400:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.25:16400;branch=z9hG4bK-F7122ACA;received=192.168.200.25
From: <sip:[email protected];user=phone>;tag=2870350082
To: <sip:[email protected];user=phone>;tag=as6c147010
Call-ID: [email protected]
CSeq: 39 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 120
Contact: <sip:[email protected]:16400;user=phone>;expires=120
Date: Wed, 25 Apr 2007 13:07:45 GMT
Content-Length: 0


---
Scheduling destruction of call '[email protected]' in 15000 ms
12 headers, 3 lines
Reliably Transmitting (no NAT) to 192.168.200.25:16396:
NOTIFY sip:[email protected]:16396;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK397322a6;rport
From: "asterisk" <sip:[email protected]>;tag=as1506a1dd
To: <sip:[email protected]:16396;user=phone>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 94

Messages-Waiting: no
Message-Account: sip:[email protected]
Voice-Message: 0/0 (0/0)
Ich denke das es an den Ports liegt, da jedoch wir uns im gleichen LAN-Segment befinden sollte das kein Problem sein
Code: 
Name/username          Host                 Dyn Nat ACL Port     Status
602/602                    192.168.200.25   D               16400    OK (3 ms)
742/742                    192.168.200.25   D               16396    OK (2 ms)

hat jemand da eine Lösung, oder Ansatz zum suchen?
 
Was bitte sollen die beiden SIP messages die Du da oben gepostet hast, mit dem beschriebenen Problem zu tun haben? Da ist doch kein relevanter Inhalt bezüglich eines Gesprächsaufbaus drin.
 
hier der erweiterte output der CLI
Code: 
Via: SIP/2.0/UDP 192.168.200.25:16400;branch=z9hG4bK-F7122AE5
From: <sip:[email protected];user=phone>;tag=2870350167
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 66 INVITE
Contact: <sip:[email protected]:16400;user=phone>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Content-Length: 204
Content-Type: application/sdp
Max-Forwards: 70
Supported: replaces,sec-agree,answermode
User-Agent: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])
P-Preferred-Identity: <sip:[email protected]:16400;user=phone>
Remote-Party-ID: <sip:[email protected]:16400;user=phone>;party=calling;screen=no;privacy=off

v=0
o=- 0 0 IN IP4 192.168.200.25
s=Audio
c=IN IP4 192.168.200.25
t=0 0
m=audio 16508 RTP/AVP 8 0 4 18 2 101
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:60

--- (15 headers 10 lines) ---
Using INVITE request as basis request - [email protected]
Sending to 192.168.200.25 : 16400 (non-NAT)
Found user '602'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 101
Peer audio RTP is at port 192.168.200.25:16508
Peer video RTP is at port 192.168.200.25:65535
Found description format G726-32
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x11d (g723|ulaw|alaw|g726|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 742 in fromsip (domain 192.168.200.20)
list_route: hop: <sip:[email protected]:16400;user=phone>
Transmitting (no NAT) to 192.168.200.25:16400:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.25:16400;branch=z9hG4bK-F7122AE5;received=192.168.200.25
From: <sip:[email protected];user=phone>;tag=2870350167
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 66 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
We're at 192.168.200.20 port 11748
Video is at 192.168.200.20 port 17392
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.200.25:16400:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.25:16400;branch=z9hG4bK-F7122AE5;received=192.168.200.25
From: <sip:[email protected];user=phone>;tag=2870350167
To: <sip:[email protected];user=phone>;tag=as276482fd
Call-ID: [email protected]
CSeq: 66 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 29694 29694 IN IP4 192.168.200.20
s=session
c=IN IP4 192.168.200.20
t=0 0
m=audio 11748 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
asterisk*CLI>
<-- SIP read from 192.168.200.25:16400:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.25:16400;branch=z9hG4bK-F7122AE5
From: <sip:[email protected];user=phone>;tag=2870350167
To: <sip:[email protected];user=phone>;tag=as276482fd
Call-ID: [email protected]
CSeq: 66 ACK
Contact: <sip:[email protected]:16400;user=phone>
Max-Forwards: 70


--- (8 headers 0 lines) ---
asterisk*CLI>
<-- SIP read from 192.168.200.25:16400:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.25:16400;branch=z9hG4bK-F7122AE6
From: <sip:[email protected];user=phone>;tag=2870350167
To: <sip:[email protected];user=phone>;tag=as276482fd
Call-ID: [email protected]
CSeq: 67 INVITE
Contact: <sip:[email protected]:16400;user=phone>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Content-Length: 190
Content-Type: application/sdp
Max-Forwards: 70
Supported: replaces,sec-agree,answermode
User-Agent: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])
P-Preferred-Identity: <sip:[email protected]:16400;user=phone>
Remote-Party-ID: <sip:[email protected]:16400;user=phone>;party=connected;screen=no;privacy=off

v=0
o=- 0 0 IN IP4 192.168.200.25
s=Audio
c=IN IP4 127.0.0.1
t=0 0
m=audio 16508 RTP/AVP 0 101
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30

--- (15 headers 10 lines) ---
Using INVITE request as basis request - [email protected]
Sending to 192.168.200.25 : 16400 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 127.0.0.1:16508
Peer video RTP is at port 127.0.0.1:65535
Found description format G726-32
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x14 (ulaw|g726)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
We're at 192.168.200.20 port 11748
Video is at 192.168.200.20 port 17392
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.200.25:16400:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.25:16400;branch=z9hG4bK-F7122AE6;received=192.168.200.25
From: <sip:[email protected];user=phone>;tag=2870350167
To: <sip:[email protected];user=phone>;tag=as276482fd
Call-ID: [email protected]
CSeq: 67 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 29694 29695 IN IP4 192.168.200.20
s=session
c=IN IP4 192.168.200.20
t=0 0
m=audio 11748 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
asterisk*CLI>
<-- SIP read from 192.168.200.25:16400:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.25:16400;branch=z9hG4bK-F7122AE6
From: <sip:[email protected];user=phone>;tag=2870350167
To: <sip:[email protected];user=phone>;tag=as276482fd
Call-ID: [email protected]
CSeq: 67 ACK
Contact: <sip:[email protected]:16400;user=phone>
Max-Forwards: 70


--- (8 headers 0 lines) ---
We're at 192.168.200.20 port 11178
Video is at 192.168.200.20 port 10390
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
13 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.200.25:16396:
INVITE sip:[email protected]:16396;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK1f743dc0;rport
From: "Saalfrank, Michael" <sip:[email protected]>;tag=as68c6ad70
To: <sip:[email protected]:16396;user=phone>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 25 Apr 2007 13:31:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 29694 29694 IN IP4 192.168.200.20
s=session
c=IN IP4 192.168.200.20
t=0 0
m=audio 11178 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 - - - -

---
asterisk*CLI>
<-- SIP read from 192.168.200.25:16396:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK1f743dc0;rport
From: "Saalfrank, Michael" <sip:[email protected]>;tag=as68c6ad70
To: <sip:[email protected]:16396;user=phone>;tag=2870350168
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Server: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])


--- (8 headers 0 lines) ---
asterisk*CLI>
<-- SIP read from 192.168.200.25:16396:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK1f743dc0;rport
From: "Saalfrank, Michael" <sip:[email protected]>;tag=as68c6ad70
To: <sip:[email protected]:16396;user=phone>;tag=2870350168
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:16396;user=phone>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Server: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])
P-Preferred-Identity: <sip:[email protected]:16396;user=phone>
Remote-Party-ID: <sip:[email protected]:16396;user=phone>;party=called;screen=no;privacy=off


--- (11 headers 0 lines) ---
 
cool...

jetzt haben wir zumindest mal ein SIP Log bis zum Klingeln der Gegenstelle. Aber danach wird es ja erst interessant - wenn versucht wird, die Audioverbindung aufzubauen. Vielleicht gelingt es ja in einem dritten Versuch, ein brauchbares SIP Log hier zu posten :-Ö
 
Code: 
Asterisk 1.2.17, Copyright (C) 1999 - 2006 Digium, Inc. and others.

Created by Mark Spencer <[email protected]>

Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.

This is free software, with components licensed under the GNU General Public

License version 2 and other licenses; you are welcome to redistribute it under

certain conditions. Type 'show license' for details.

=========================================================================

Connected to Asterisk 1.2.17 currently running on asterisk (pid = 29694)
asterisk*CLI> 
<-- SIP read from 192.168.200.25:16396: 
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.25:16396;branch=z9hG4bK-F7122F30
From: <sip:[email protected];user=phone>;tag=2870353964
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1165 INVITE
Contact: <sip:[email protected]:16396;user=phone>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Content-Length: 204
Content-Type: application/sdp
Max-Forwards: 70
Supported: replaces,sec-agree,answermode
User-Agent: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])
P-Preferred-Identity: <sip:[email protected]:16396;user=phone>
Remote-Party-ID: <sip:[email protected]:16396;user=phone>;party=calling;screen=no;privacy=off

v=0
o=- 0 0 IN IP4 192.168.200.25
s=Audio
c=IN IP4 192.168.200.25
t=0 0
m=audio 16568 RTP/AVP 8 0 4 18 2 101
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:60

 --- (15 headers 10 lines) ---
 
asterisk*CLI> 
Using INVITE request as basis request - [email protected]
 Sending to 192.168.200.25 : 16396 (non-NAT)
 Found user '742'
 Found RTP audio format 8
 Found RTP audio format 0
 Found RTP audio format 4
 Found RTP audio format 18
 Found RTP audio format 2
 Found RTP audio format 101
 Peer audio RTP is at port 192.168.200.25:16568
 Peer video RTP is at port 192.168.200.25:65535
 Found description format G726-32
 Found description format telephone-event
 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x11d (g723|ulaw|alaw|g726|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 602 in fromsip (domain 192.168.200.20)
 list_route: hop: <sip:[email protected]:16396;user=phone>
 Transmitting (no NAT) to 192.168.200.25:16396:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.25:16396;branch=z9hG4bK-F7122F30;received=192.168.200.25
From: <sip:[email protected];user=phone>;tag=2870353964
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1165 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
 
asterisk*CLI> 
We're at 192.168.200.20 port 16286
 
asterisk*CLI> 
Video is at 192.168.200.20 port 19708
 
asterisk*CLI> 
Adding codec 0x4 (ulaw) to SDP
 
asterisk*CLI> 
Adding codec 0x8 (alaw) to SDP
 
asterisk*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
 
asterisk*CLI> 
Reliably Transmitting (no NAT) to 192.168.200.25:16396:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.25:16396;branch=z9hG4bK-F7122F30;received=192.168.200.25
From: <sip:[email protected];user=phone>;tag=2870353964
To: <sip:[email protected];user=phone>;tag=as2b64f154
Call-ID: [email protected]
CSeq: 1165 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
C
asterisk*CLI> 
ontent-Length: 244

v=0
o=root 29694 29694 IN IP4 192.168.200.20
s=session
c=IN IP4 192.168.200.20
t=0 0
m=audio 16286 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
 
asterisk*CLI> 
<-- SIP read from 192.168.200.25:16396: 
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.25:16396;branch=z9hG4bK-F7122F30
From: <sip:[email protected];user=phone>;tag=2870353964
To: <sip:[email protected];user=phone>;tag=as2b64f154
Call-ID: [email protected]
CSeq: 1165 ACK
Contact: <sip:[email protected]:16396;user=phone>
Max-Forwards: 70


 
asterisk*CLI> 
--- (8 headers 0 lines) ---
 
asterisk*CLI> 
<-- SIP read from 192.168.200.25:16396: 
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.25:16396;branch=z9hG4bK-F7122F31
From: <sip:[email protected];user=phone>;tag=2870353964
To: <sip:[email protected];user=phone>;tag=as2b64f154
Call-ID: [email protected]
CSeq: 1166 INVITE
Contact: <sip:[email protected]:16396;user=phone>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Content-Length: 190
Content-Type: application/sdp
Max-Forwards: 70
Supported: replaces,sec-agree,answermode
User-Agent: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])
P-Preferred-Identity: <sip:[email protected]:16396;user=phone>
Remote-Party-ID: <sip:[email protected]:16396;user=phone>;party=connected;screen=no;privacy=off

v=0
o=- 0 0 IN IP4 192.168.200.25
s=Audio
c=IN IP4 127.0.0.1
t=0 0
m=audio 16568 RTP/AVP 0 101
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30

 
asterisk*CLI> 
--- (15 headers 10 lines) ---
 
asterisk*CLI> 
Using INVITE request as basis request - [email protected]
 
asterisk*CLI> 
Sending to 192.168.200.25 : 16396 (non-NAT)
 
asterisk*CLI> 
Found RTP audio format 0
 
asterisk*CLI> 
Found RTP audio format 101
 
asterisk*CLI> 
Peer audio RTP is at port 127.0.0.1:16568
 
asterisk*CLI> 
Peer video RTP is at port 127.0.0.1:65535
 
asterisk*CLI> 
Found description format G726-32
 
asterisk*CLI> 
Found description format telephone-event
 
asterisk*CLI> 
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x14 (ulaw|g726)/video=0x0 (nothing), combined - 0x4 (ulaw)
 
asterisk*CLI> 
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
 
asterisk*CLI> 
We're at 192.168.200.20 port 16286
 
Kasterisk*CLI> 
Video is at 192.168.200.20 port 19708
 
Kasterisk*CLI> 
Adding codec 0x4 (ulaw) to SDP
 
Kasterisk*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
 
asterisk*CLI> 
Reliably Transmitting (no NAT) to 192.168.200.25:16396:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.25:16396;branch=z9hG4bK-F7122F31;received=192.168.200.25
From: <sip:[email protected];user=phone>;tag=2870353964
To: <sip:[email protected];user=phone>;tag=as2b64f154
Call-ID: [email protected]
CSeq: 1166 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
C
asterisk*CLI> 
ontent-Length: 220

v=0
o=root 29694 29695 IN IP4 192.168.200.20
s=session
c=IN IP4 192.168.200.20
t=0 0
m=audio 16286 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
 
asterisk*CLI> 
<-- SIP read from 192.168.200.25:16396: 
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.25:16396;branch=z9hG4bK-F7122F31
From: <sip:[email protected];user=phone>;tag=2870353964
To: <sip:[email protected];user=phone>;tag=as2b64f154
Call-ID: [email protected]
CSeq: 1166 ACK
Contact: <sip:[email protected]:16396;user=phone>
Max-Forwards: 70


 
asterisk*CLI> 
--- (8 headers 0 lines) ---
 
asterisk*CLI> 
We're at 192.168.200.20 port 12940
 
asterisk*CLI> 
Video is at 192.168.200.20 port 16724
 
asterisk*CLI> 
Adding codec 0x4 (ulaw) to SDP
 
asterisk*CLI> 
Adding codec 0x2 (gsm) to SDP
 
asterisk*CLI> 
Adding codec 0x8 (alaw) to SDP
 
asterisk*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP
 
asterisk*CLI> 
13 headers, 12 lines
 
asterisk*CLI> 
Reliably Transmitting (no NAT) to 192.168.200.25:16400:
INVITE sip:[email protected]:16400;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK4a32b61c;rport
From: "742" <sip:[email protected]>;tag=as79c984d5
To: <sip:[email protected]:16400;user=phone>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 26 Apr 2007 07:26:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 29694 29694 IN IP4 192.168.200.20
s=session
c=IN IP4 192.168.200.20
t=0 0
m=audio 12940 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 - - - -

---
 
asterisk*CLI> 
<-- SIP read from 192.168.200.25:16400: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK4a32b61c;rport
From: "742" <sip:[email protected]>;tag=as79c984d5
To: <sip:[email protected]:16400;user=phone>;tag=2870353965
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Server: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])


 
asterisk*CLI> 
--- (8 headers 0 lines) ---
 
asterisk*CLI> 
<-- SIP read from 192.168.200.25:16400: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK4a32b61c;rport
From: "742" <sip:[email protected]>;tag=as79c984d5
To: <sip:[email protected]:16400;user=phone>;tag=2870353965
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:16400;user=phone>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Server: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])
P-Preferred-Identity: <sip:[email protected]:16400;user=phone>
Remote-Party-ID: <sip:[email protected]:16400;user=phone>;party=called;screen=no;privacy=off


 
asterisk*CLI> 
--- (11 headers 0 lines) ---
 
asterisk*CLI> 
<-- SIP read from 192.168.200.25:16400: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK4a32b61c;rport
From: "742" <sip:[email protected]>;tag=as79c984d5
To: <sip:[email protected]:16400;user=phone>;tag=2870353965
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:16400;user=phone>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Content-Length: 208
Content-Type: application/sdp
Server: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])
P-Preferred-Identity: <sip:[email protected]:16400;user=phone>
Remote-Party-ID: <sip:[email protected]:16400;user=phone>;party=called;screen=no;privacy=off

v=0
o=root 29694 29694 IN IP4 192.168.200.20
s=session
c=IN IP4 192.168.200.25
t=0 0
m=audio 16572 RTP/AVP 8 101
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:60

 
asterisk*CLI> 
--- (13 headers 10 lines) ---
 
asterisk*CLI> 
Found RTP audio format 8
 
asterisk*CLI> 
Found RTP audio format 101
 
asterisk*CLI> 
Peer audio RTP is at port 192.168.200.25:16572
 
asterisk*CLI> 
Peer video RTP is at port 192.168.200.25:65535
 
asterisk*CLI> 
Found description format G726-32
 
asterisk*CLI> 
Found description format telephone-event
 
asterisk*CLI> 
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x18 (alaw|g726)/video=0x0 (nothing), combined - 0x8 (alaw)
 
asterisk*CLI> 
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
 
asterisk*CLI> 
list_route: hop: <sip:[email protected]:16400;user=phone>
 
asterisk*CLI> 
set_destination: Parsing <sip:[email protected]:16400;user=phone> for address/port to send to
 
asterisk*CLI> 
set_destination: set destination to 192.168.200.25, port 16400
 
asterisk*CLI> 
Transmitting (no NAT) to 192.168.200.25:16400:
ACK sip:[email protected]:16400;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK18cc4f06;rport
From: "742" <sip:[email protected]>;tag=as79c984d5
To: <sip:[email protected]:16400;user=phone>;tag=2870353965
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
 
asterisk*CLI> 
<-- SIP read from 192.168.200.25:16396: 
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.25:16396;branch=z9hG4bK-F7122F32
From: <sip:[email protected];user=phone>;tag=2870353964
To: <sip:[email protected];user=phone>;tag=as2b64f154
Call-ID: [email protected]
CSeq: 1167 BYE
Contact: <sip:[email protected]:16396;user=phone>
Max-Forwards: 70


 
asterisk*CLI> 
--- (8 headers 0 lines) ---
 
asterisk*CLI> 
Sending to 192.168.200.25 : 16396 (non-NAT)
 
asterisk*CLI> 
Transmitting (no NAT) to 192.168.200.25:16396:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.25:16396;branch=z9hG4bK-F7122F32;received=192.168.200.25
From: <sip:[email protected];user=phone>;tag=2870353964
To: <sip:[email protected];user=phone>;tag=as2b64f154
Call-ID: [email protected]
CSeq: 1167 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
 
asterisk*CLI> 
Scheduling destruction of call '[email protected]' in 32000 ms
 
asterisk*CLI> 
set_destination: Parsing <sip:[email protected]:16400;user=phone> for address/port to send to
 
asterisk*CLI> 
set_destination: set destination to 192.168.200.25, port 16400
 
asterisk*CLI> 
Reliably Transmitting (no NAT) to 192.168.200.25:16400:
BYE sip:[email protected]:16400;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK3b538d54;rport
From: "742" <sip:[email protected]>;tag=as79c984d5
To: <sip:[email protected]:16400;user=phone>;tag=2870353965
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
 
asterisk*CLI> 
<-- SIP read from 192.168.200.25:16400: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK3b538d54;rport
From: "742" <sip:[email protected]>;tag=as79c984d5
To: <sip:[email protected]:16400;user=phone>;tag=2870353965
Call-ID: [email protected]
CSeq: 103 BYE
Server: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])


 
asterisk*CLI> 
--- (7 headers 0 lines) ---
 
asterisk*CLI> 
Destroying call '[email protected]'
 
asterisk*CLI> qui
<-- SIP read from 192.168.200.25:16400: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.20:5060;branch=z9hG4bK3b538d54;rport
From: "742" <sip:[email protected]>;tag=as79c984d5
To: <sip:[email protected]:16400;user=phone>;tag=2870353965
Call-ID: [email protected]
CSeq: 103 BYE
Server: (KIRK Wireless Server 600/3/6.00 dvl [07-60400.78])


 
asterisk*CLI> qui
--- (7 headers 0 lines) ---
 
asterisk*CLI> qui
Destroying call '[email protected]'
 
asterisk*CLI> quit
 
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