root@asterisk:/etc/asterisk# asterisk -r
Asterisk 1.8.23.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.23.0 currently running on asterisk (pid = 2579)
asterisk*CLI> sip set debug on
SIP Debugging enabled
asterisk*CLI> exit
root@asterisk:/etc/asterisk# asterisk -r
Asterisk 1.8.23.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
#Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.23.0 currently running on asterisk (pid = 2579)
<--- SIP read from UDP:192.168.178.1:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.1:5060;branch=z9hG4bKD2996B432C68C4CC
From: "***NAME IN FRITZBOX TELEFONBUCH***" <sip:***ANRUFER-NUMMER***@fritz.fonwlan.box>;tag=3CF0B21CFA0A4C08
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 44 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7270 v3 (UI) 74.05.50 (Jan 14 2013)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 419
v=0
o=user 1659763 1659763 IN IP4 192.168.178.1
s=call
c=IN IP4 192.168.178.1
t=0 0
m=audio 7082 RTP/AVP 9 8 0 2 102 100 99 97 120 121 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:120 PCMA/16000
a=rtpmap:121 PCMU/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:7083
<------------->
--- (17 headers 18 lines) ---
Sending to 192.168.178.1:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '621' for '***Anrufer-Nummer***' from 192.168.178.1:5060
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 99
Found RTP audio format 97
Found RTP audio format 120
Found RTP audio format 121
Found RTP audio format 101
Found audio description format G726-32 for ID 2
Found audio description format G726-32 for ID 102
Found unknown media description format G726-40 for ID 100
Found unknown media description format G726-24 for ID 99
Found audio description format iLBC for ID 97
Found unknown media description format PCMA for ID 120
Found unknown media description format PCMU for ID 121
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x1c0c (ulaw|alaw|g726|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.1:7082
Looking for 621 in default (domain 192.168.178.30)
<--- Reliably Transmitting (NAT) to 192.168.178.1:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.178.1:5060;branch=z9hG4bKD2996B432C68C4CC;received=192.168.178.1;rport=5060
From: "***NAME IN FRITZBOX TELEFONBUCH***" <sip:***ANRUFER-NUMMER***@fritz.fonwlan.box>;tag=3CF0B21CFA0A4C08
To: <sip:[email protected]:5060>;tag=as241c4d44
Call-ID: [email protected]
CSeq: 44 INVITE
Server: Asterisk PBX 1.8.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.178.1:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.1:5060;branch=z9hG4bKD2996B432C68C4CC
From: "***NAME IN FRITZBOX TELEFONBUCH***" <sip:***ANRUFER-NUMMER***@fritz.fonwlan.box>;tag=3CF0B21CFA0A4C08
To: <sip:[email protected]:5060>;tag=as241c4d44
Call-ID: [email protected]
CSeq: 44 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 7270 v3 (UI) 74.05.50 (Jan 14 2013)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.178.1:5060:
REGISTER sip:fritz.fonwlan.box SIP/2.0
Via: SIP/2.0/UDP 192.168.178.30:5060;branch=z9hG4bK5acf099b;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as26aa594b
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 116 REGISTER
User-Agent: Asterisk PBX 1.8.23.0
Authorization: Digest username="621", realm="fritz.box", algorithm=MD5, uri="sip:fritz.fonwlan.box", nonce="28163F9760154D27", response="b1acd17c072f3c27ba497fffd156747e"
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0
---
<--- SIP read from UDP:192.168.178.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.30:5060;branch=z9hG4bK5acf099b;rport=5060
From: <sip:[email protected]>;tag=as26aa594b
To: <sip:[email protected]>;tag=C299E2A20E73A32A
Call-ID: [email protected]
CSeq: 116 REGISTER
WWW-Authenticate: Digest realm="fritz.box", nonce="3DB101D68641980B"
User-Agent: AVM FRITZ!Box Fon WLAN 7270 v3 (UI) 74.05.50 (Jan 14 2013)
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Responding to challenge, registration to domain/host name fritz.fonwlan.box
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.178.1:5060:
REGISTER sip:fritz.fonwlan.box SIP/2.0
Via: SIP/2.0/UDP 192.168.178.30:5060;branch=z9hG4bK080ae253;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as4996ad71
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 117 REGISTER
User-Agent: Asterisk PBX 1.8.23.0
Authorization: Digest username="621", realm="fritz.box", algorithm=MD5, uri="sip:fritz.fonwlan.box", nonce="3DB101D68641980B", response="6db6241cce2ee8c0e69d9e01e6b1ad32"
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0
---
<--- SIP read from UDP:192.168.178.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.30:5060;branch=z9hG4bK080ae253;rport=5060
From: <sip:[email protected]>;tag=as4996ad71
To: <sip:[email protected]>;tag=22C1E316CAC7263D
Call-ID: [email protected]
CSeq: 117 REGISTER
Contact: <sip:[email protected]:5060>;expires=300
User-Agent: AVM FRITZ!Box Fon WLAN 7270 v3 (UI) 74.05.50 (Jan 14 2013)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer,reg
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '[email protected]' Method: REGISTER
asterisk*CLI>