[Gelöst] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) -- fb7390

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andreas96k

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Hallo,


Ich bin gerade dabei Asterisk auf dem svn Tag 11685 zu testen (vom 10.02.2014)
Bei den Tests bin ich auf mehrere Ungereimtheiten gestoßen. u.a.:
a) Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
b) Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
c) Das System verhält sich nach jedem boot anders.


Basierend aus meinen Erfahreungen aus
"korrekte konfiguration von Aterisk asterisk-11.8.0-rc1 (svn tag 11641) auf fb7390 <http://www.ip-phone-forum.de/showthread.php?t=266864>"

habe ich folgende freetz Builds erstellt:
a) USB-root host mit
- Level of user competence
= expert
- replace kernel
- removal patches ->
= AHA
= FHEM
- other patches
= ext2
packages -> packages ->
= bftpd
= USB root
b) Asterisk mit folgenden weiteren Einstellungen
- packages -> packages ->
= AVM-forewarding (AVM-firewall ist scheinbar nicht mehr unterstützt)
= Midnight Commander
- packages -> unstable ->
= asterisk (default einstellungen)

Im Einsatz habe ich vier Komponenten:
1. primäre fb7390 (Verbindung ins Internet, zur Verfügung Stellung der "externen Leitungen"
2. asterisk fb7390 (mit oben beschriebenem Build)
3. PhonerLite 1.95 als SIP Telefon an Asterisk
4. DECT Telefon am primären fb7390

Die primäre fb7390 hat:
- zwei konfigurierte SIP IP-Telefone (**620 & **621)

Die Asterisk fb7390 hat:
- zwei externe Telefonnummern (620 & 621 / verbunden mit der primären fb7390
- einen AB der auf einer Asterisk Nebenstellen (2001) reagiert
- Das asterisk in dieser fb7390 verwendet 620 & 621 als externe Leitungen und stellt zwei Nebenstellen (2000 & 2001) zur Verfügung
- das asterisk leitet 620 an 2000 und 621 an 2001 weiter.
- Im AVM-Forewarding sind die folgenden Ports weitergeleitet (je TCP und UDP; 5061 (asterisk) & 10000-20000 (s. RTP.conf)

PhonerLite:
- ist mit der Nebenstelle 2000 verbunden


a) Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
Nach jedem Reboot ist das Unterverzeichnis 'cdr-csv' im Verzeichnis '/var/log/asterisk/' gelöscht.
Dadurch gibt es immer eine Fehlermeldung wenn man mitder Konsole Verbunden ist.
Weiß jemand, wie man das Unterverzeichnis 'reboot sicher' erstellen kann?


b) Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Wie es scheint meldet sich der AB der asterisk fb7390 nicht immer sauber am asterisk an.
Ein reboot kann dieses Problem lösen.
Durch ein weiteres reboot kann es allerdings wiederkommen.
Dies Problem stellt sich nie beim PhonerLite ein. Dieser meldet sich immer korrekt am asterisk an.
Ich hatte gehofft, das eine Freigabe des asterisk ports 5061 und der ports in rtp.conf mich einer Lösung näher bringen. Leider ohen Erfolg.
Ich gehe davon aus, dass die Firewall nicht korrekt konfiguriert ist. Weiß jemand, wie man dieses Problem umgehen kann?



c) Das System verhält sich nach jedem boot anders.
Nach jedem Boot startet das System anders und verhält sich anders (identische Konfiguration).
- Einmal verbindet sich der AB der Fritzbox, ein andernmal nicht
- Einmal kann ich vom DECT direkt über asterisk PhonerLite anrufen,ein ander mal muss ich erst auf PhonerLite versuchen eine Verbindung aufzubauen.
In einem digitalen System tritt so etwas normalerweise nur bei Laufzeitproblemen aus.
Kennt jemand eine Möglichkeit, wie ich das starten der verschiedenen Systemen (insbesondere vom asterisk steuern / verzögern kann?



sip.conf
Code: 
[global]
;session-timer=refuse
;srvlookup=yes

[general]
language=de
disallow=all
allow=ulaw
allow=alaw
tcpenable=yes
tcpbindaddr=0.0.0.0:5061
bindaddr = 0.0.0.0:5061
nat=no
context = fritzbox
canreinvite=no
alwaysauthreject=yes
allowguest=no
;bindport=5061			; UDP Port to bind to (SIP standard port is 5060)
;bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)

register => 620:[email protected]/620
register => 621:[email protected]/621

[2000] 
type=friend
context=localphone
secret=1234 
host=dynamic
nat=no
qualify=no

[2001] 
type=friend
contect=localphone
secret=1234
host=dynamic
nat=no
qualify=no

[620]
type=peer
defaultuser=620
fromuser=620
secret=620
host=192.168.140.29
insecure=port,invite
canreinvite=no
dtmfmode=rfc2833
qualify=no
nat=no
context=fritzbox

[621]
type=peer
defaultuser=621
fromuser=621
secret=621
host=192.168.140.29
insecure=port,invite
canreinvite=no
dtmfmode=rfc2833
qualify=no
nat=no
context=fritzbox


extension.conf
Code: 
[sonstige]

[localphone]
exten => 1001,1,Answer()
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Hangup()

exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)

exten => 2999,1,VoiceMailMain(${CALLERID(num),s}

exten => _0[1-9].,1,Dial(SIP/${EXTEN}@sips) ;ext-sip-account

[fritzbox]
exten => 620,1,Dial(SIP/2000)
exten => 621,1,Dial(SIP/2001)


rtp.conf (Auszug)
Code: 
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;


traces A nach einem reboot (u.a. /var/log/asterisk//cdr-csv//Master.csv)
Code: 
192.168.140.29 login: root
Password:
   __  _   __  __ ___ __
  |__ |_) |__ |__  |   /
  |   |\  |__ |__  |  /_

   The fun has just begun ...


BusyBox v1.22.1 (2014-02-04 21:41:53 CET) built-in shell (ash)
Enter 'help' for a list of built-in commands.

root@192:/var/mod/root# asterisk -rvvv
Asterisk 11.8.0-rc1, Copyright (C) 1999 - 2013 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 11.8.0-rc1 currently running on 192 (pid = 2000)
    -- Remote UNIX connection
192*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:192.168.140.21:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK008c56b55893e311ab8a08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 58 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:[email protected]>
Content-Length: 418

v=0
o=- 4143207555 0 IN IP4 192.168.140.41
s=SIPPER for PhonerLite
c=IN IP4 192.168.140.41
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:2854731364
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Sending to 192.168.140.21:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '2000' for '2000' from 192.168.140.21:5060

<--- Reliably Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK008c56b55893e311ab8a08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>;tag=as2e13571e
Call-ID: [email protected]
CSeq: 58 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="163734a7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.140.21:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK008c56b55893e311ab8a08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>;tag=as2e13571e
Call-ID: [email protected]
CSeq: 58 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.140.21:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK008c56b55893e311ab8b08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 59 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="163734a7", uri="sip:[email protected]", response="1a8a25bd09cba6e97b2ad66139d94289", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:[email protected]>
Content-Length: 418

v=0
o=- 4143207555 0 IN IP4 192.168.140.21
s=SIPPER for PhonerLite
c=IN IP4 192.168.140.21
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:2854731364
a=sendrecv
<------------->
--- (15 headers 18 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '2000' for '2000' from 192.168.140.21:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|g726|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.21:5062
Looking for 2001 in localphone (domain 192.168.140.29)
list_route: hop: <sip:[email protected]:5060>

<--- Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK008c56b55893e311ab8b08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 59 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [2001@localphone:1] Dial("SIP/2000-00000006", "SIP/2001") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 12686
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.140.29:5060:
INVITE sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK1c3fdcfe;rport
Max-Forwards: 70
From: "2000" <sip:[email protected]:5061>;tag=as43a2cec2
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.0-rc1
Date: Thu, 13 Feb 2014 20:11:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 567222338 567222338 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 12686 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 - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.140.29:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK1c3fdcfe;rport=5061
From: "2000" <sip:[email protected]:5061>;tag=as43a2cec2
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
Call-ID: [email protected]:5061
CSeq: 102 INVITE
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Called SIP/2001

<--- SIP read from UDP:192.168.140.29:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK1c3fdcfe;rport=5061
From: "2000" <sip:[email protected]:5061>;tag=as43a2cec2
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=2C1A360784C3D6B2
Call-ID: [email protected]:5061
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
    -- SIP/2001-00000007 is ringing

<--- Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK008c56b55893e311ab8b08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>;tag=as5fbc9888
Call-ID: [email protected]
CSeq: 59 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.140.29:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK1c3fdcfe;rport=5061
From: "2000" <sip:[email protected]:5061>;tag=as43a2cec2
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=2C1A360784C3D6B2
Call-ID: [email protected]:5061
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Supported: 100rel,replaces
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: 275

v=0
o=user 11831696 11831696 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:7079
a=ptime:30
<------------->
--- (15 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.29:7078
list_route: hop: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
set_destination: Parsing <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305> for address/port to send to
set_destination: set destination to 192.168.140.29:5060
Transmitting (NAT) to 192.168.140.29:5060:
ACK sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK25b9c88d;rport
Max-Forwards: 70
From: "2000" <sip:[email protected]:5061>;tag=as43a2cec2
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=2C1A360784C3D6B2
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.0-rc1
Content-Length: 0


---
    -- SIP/2001-00000007 answered SIP/2000-00000006
Audio is at 13052
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK008c56b55893e311ab8b08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>;tag=as5fbc9888
Call-ID: [email protected]
CSeq: 59 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5061>
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 718488107 718488107 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 13052 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 - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Locally bridging SIP/2000-00000006 and SIP/2001-00000007

<--- SIP read from UDP:192.168.140.21:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK807c51b85893e311ab8b08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>;tag=as5fbc9888
Call-ID: [email protected]
CSeq: 59 ACK
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="163734a7", uri="sip:[email protected]:5061", response="731cfd2a27e20ddd0e21b5a19b6855b1", algorithm=MD5
Max-Forwards: 70
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.140.21:5060 --->
BYE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK006d4cbb5893e311ab8b08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>;tag=as5fbc9888
Call-ID: [email protected]
CSeq: 60 BYE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="163734a7", uri="sip:[email protected]:5061", response="6775686053a66d21db39dc4c0dfdeaca", algorithm=MD5
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK006d4cbb5893e311ab8b08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=1867644136
To: <sip:[email protected]>;tag=as5fbc9888
Call-ID: [email protected]
CSeq: 60 BYE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Feb 13 20:11:48] ERROR[3540][C-00000003]: cdr_csv.c:305 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
Scheduling destruction of SIP dialog '[email protected]:5061' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305> for address/port to send to
set_destination: set destination to 192.168.140.29:5060
Reliably Transmitting (NAT) to 192.168.140.29:5060:
BYE sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK196f3428;rport
Max-Forwards: 70
From: "2000" <sip:[email protected]:5061>;tag=as43a2cec2
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=2C1A360784C3D6B2
Call-ID: [email protected]:5061
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.8.0-rc1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.140.29:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK196f3428;rport=5061
From: "2000" <sip:[email protected]:5061>;tag=as43a2cec2
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=2C1A360784C3D6B2
Call-ID: [email protected]:5061
CSeq: 103 BYE
X-RTP-Stat: CS=0;PS=83;ES=173;OS=19920;SP=0/0;SO=0;QS=-;PR=256;ER=260;OR=40960;CR=0;SR=0;QR=-;PL=0,0;BL=0;LS=0;RB=0/0;SB=-/-;EN=PCMU;DE=PCMU;JI=25,0;DL=0,0,0;IP=192.168.140.29:7078,192.168.140.29:12686
X-RTP-Stat-Add: DQ=6;DSS=0;DS=0;PLCS=96;JS=1
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
  == Spawn extension (localphone, 2001, 1) exited non-zero on 'SIP/2000-00000006'

<--- SIP read from UDP:192.168.140.29:5060 --->
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bKC13371A6FB898E25
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=3B0E345E3FD84BFC
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
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: 423

v=0
o=user 15411011 15411011 IN IP4 192.168.140.29
s=call
c=IN IP4 192.168.140.29
t=0 0
m=audio 7086 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:7087
<------------->
--- (17 headers 18 lines) ---
Sending to 192.168.140.29:5060 (NAT)
Sending to 192.168.140.29:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '620' for '***123#*#**610' from 192.168.140.29:5060
  == Using SIP RTP CoS mark 5
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 - (ulaw|alaw), peer - audio=(ulaw|alaw|g726|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.29:7086
Looking for 620 in fritzbox (domain 192.168.140.29)
list_route: hop: <sip:[email protected]>

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bKC13371A6FB898E25;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=3B0E345E3FD84BFC
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 3 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [620@fritzbox:1] Dial("SIP/620-00000008", "SIP/2000") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14184
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.140.41:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK79bd0734;rport
Max-Forwards: 70
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]:5061>;tag=as536825f3
To: <sip:[email protected]:5060>
Contact: <sip:***123%23*%23**[email protected]:5061>
Call-ID: [email protected]:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.0-rc1
Date: Thu, 13 Feb 2014 20:11:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 294

v=0
o=root 1056737477 1056737477 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 14184 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 - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/2000

<--- SIP read from UDP:192.168.140.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK79bd0734;rport=5061
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]:5061>;tag=as536825f3
To: <sip:[email protected]:5060>
Call-ID: [email protected]:5061
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.140.21:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK79bd0734;rport=5061
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]:5061>;tag=as536825f3
To: <sip:[email protected]:5060>;tag=00f4dfbe5893e311ab8b08002700ec4a
Call-ID: [email protected]:5061
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
list_route: hop: <sip:[email protected]:5060>
    -- SIP/2000-00000009 is ringing

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bKC13371A6FB898E25;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=3B0E345E3FD84BFC
To: <sip:[email protected]:5061>;tag=as4f2511b4
Call-ID: [email protected]
CSeq: 3 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.140.29:5060 --->
CANCEL sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bKC13371A6FB898E25
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=3B0E345E3FD84BFC
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 3 CANCEL
Reason: Q.850; cause=16
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.140.29:5060 (NAT)

<--- Reliably Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bKC13371A6FB898E25;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=3B0E345E3FD84BFC
To: <sip:[email protected]:5061>;tag=as4f2511b4
Call-ID: [email protected]
CSeq: 3 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bKC13371A6FB898E25;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=3B0E345E3FD84BFC
To: <sip:[email protected]:5061>;tag=as4f2511b4
Call-ID: [email protected]
CSeq: 3 CANCEL
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]:5061' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.140.21:5060:
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK79bd0734;rport
Max-Forwards: 70
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]:5061>;tag=as536825f3
To: <sip:[email protected]:5060>
Call-ID: [email protected]:5061
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.8.0-rc1
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]:5061' in 32000 ms (Method: INVITE)
  == Spawn extension (fritzbox, 620, 1) exited non-zero on 'SIP/620-00000008'

<--- SIP read from UDP:192.168.140.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK79bd0734;rport=5061
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]:5061>;tag=as536825f3
To: <sip:[email protected]:5060>;tag=00f4dfbe5893e311ab8b08002700ec4a
Call-ID: [email protected]:5061
CSeq: 102 CANCEL
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.140.21:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK79bd0734;rport=5061
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]:5061>;tag=as536825f3
To: <sip:[email protected]:5060>;tag=00f4dfbe5893e311ab8b08002700ec4a
Call-ID: [email protected]:5061
CSeq: 102 INVITE
Contact: <sip:[email protected]:5060>
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.140.21:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK79bd0734;rport
Max-Forwards: 70
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]:5061>;tag=as536825f3
To: <sip:[email protected]:5060>;tag=00f4dfbe5893e311ab8b08002700ec4a
Contact: <sip:***123%23*%23**[email protected]:5061>
Call-ID: [email protected]:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.0-rc1
Content-Length: 0


---
[Feb 13 20:11:56] ERROR[3541][C-00000004]: cdr_csv.c:305 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
Scheduling destruction of SIP dialog '[email protected]:5061' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.140.29:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bKC13371A6FB898E25
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=3B0E345E3FD84BFC
To: <sip:[email protected]:5061>;tag=as4f2511b4
Call-ID: [email protected]
CSeq: 3 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:192.168.140.29:5060 --->
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK3ADC45D5FD312766
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=52303F50352F6E27
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
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: 421

v=0
o=user 7608357 7608357 IN IP4 192.168.140.29
s=call
c=IN IP4 192.168.140.29
t=0 0
m=audio 7078 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:7079
<------------->
--- (17 headers 18 lines) ---
Sending to 192.168.140.29:5060 (NAT)
Sending to 192.168.140.29:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '620' for '***124#*#**610' from 192.168.140.29:5060
  == Using SIP RTP CoS mark 5
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 - (ulaw|alaw), peer - audio=(ulaw|alaw|g726|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.29:7078
Looking for 621 in fritzbox (domain 192.168.140.29)
list_route: hop: <sip:[email protected]>

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK3ADC45D5FD312766;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=52303F50352F6E27
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [621@fritzbox:1] Dial("SIP/620-0000000a", "SIP/2001") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 17412
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.140.29:5060:
INVITE sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK029e41fb;rport
Max-Forwards: 70
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]:5061>;tag=as040f080b
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
Contact: <sip:***124%23*%23**[email protected]:5061>
Call-ID: [email protected]:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.0-rc1
Date: Thu, 13 Feb 2014 20:12:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 683770061 683770061 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 17412 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 - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.140.29:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK029e41fb;rport=5061
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]:5061>;tag=as040f080b
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
Call-ID: [email protected]:5061
CSeq: 102 INVITE
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Called SIP/2001

<--- SIP read from UDP:192.168.140.29:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK029e41fb;rport=5061
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]:5061>;tag=as040f080b
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=007D8B996C3A9FB7
Call-ID: [email protected]:5061
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
    -- SIP/2001-0000000b is ringing

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK3ADC45D5FD312766;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=52303F50352F6E27
To: <sip:[email protected]:5061>;tag=as14bf1901
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.140.29:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK029e41fb;rport=5061
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]:5061>;tag=as040f080b
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=007D8B996C3A9FB7
Call-ID: [email protected]:5061
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Supported: 100rel,replaces
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: 275

v=0
o=user 13570769 13570769 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 7090 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:7091
a=ptime:30
<------------->
--- (15 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.29:7090
list_route: hop: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>
set_destination: Parsing <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305> for address/port to send to
set_destination: set destination to 192.168.140.29:5060
Transmitting (NAT) to 192.168.140.29:5060:
ACK sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK4ecaf3d7;rport
Max-Forwards: 70
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]:5061>;tag=as040f080b
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=007D8B996C3A9FB7
Contact: <sip:***124%23*%23**[email protected]:5061>
Call-ID: [email protected]:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.0-rc1
Content-Length: 0


---
    -- SIP/2001-0000000b answered SIP/620-0000000a
Audio is at 13820
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK3ADC45D5FD312766;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=52303F50352F6E27
To: <sip:[email protected]:5061>;tag=as14bf1901
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5061>
Content-Type: application/sdp
Require: timer
Content-Length: 292

v=0
o=root 102701106 102701106 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 13820 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 - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Locally bridging SIP/620-0000000a and SIP/2001-0000000b

<--- SIP read from UDP:192.168.140.29:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK1EF39B65AC6C2612
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=52303F50352F6E27
To: <sip:[email protected]:5061>;tag=as14bf1901
Call-ID: [email protected]
CSeq: 2 ACK
Contact: <sip:[email protected]>
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.140.29:5060 --->
BYE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK196D12F2C36821D5
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=52303F50352F6E27
To: <sip:[email protected]:5061>;tag=as14bf1901
Call-ID: [email protected]
CSeq: 3 BYE
X-RTP-Stat: CS=185;PS=122;ES=160;OS=19520;SP=0/0;SO=0;QS=-;PR=107;ER=160;OR=25680;CR=0;SR=0;QR=-;PL=0,0;BL=0;LS=0;RB=0/0;SB=-/-;EN=PCMU;DE=PCMU;JI=39,0;DL=0,0,0;IP=192.168.140.29:7078,192.168.140.29:13820
X-RTP-Stat-Add: DQ=17;DSS=0;DS=0;PLCS=128;JS=9
Reason: Q.850; cause=16
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.140.29:5060 (NAT)
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK196D12F2C36821D5;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=52303F50352F6E27
To: <sip:[email protected]:5061>;tag=as14bf1901
Call-ID: [email protected]
CSeq: 3 BYE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Feb 13 20:12:08] ERROR[3542][C-00000005]: cdr_csv.c:305 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
Scheduling destruction of SIP dialog '[email protected]:5061' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305> for address/port to send to
set_destination: set destination to 192.168.140.29:5060
Reliably Transmitting (NAT) to 192.168.140.29:5060:
BYE sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK6b0e8f06;rport
Max-Forwards: 70
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]:5061>;tag=as040f080b
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=007D8B996C3A9FB7
Call-ID: [email protected]:5061
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.8.0-rc1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (fritzbox, 621, 1) exited non-zero on 'SIP/620-0000000a'

<--- SIP read from UDP:192.168.140.29:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK6b0e8f06;rport=5061
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]:5061>;tag=as040f080b
To: <sip:[email protected];uniq=E5ADB09FD7774E0CA31E33B5F3305>;tag=007D8B996C3A9FB7
Call-ID: [email protected]:5061
CSeq: 103 BYE
X-RTP-Stat: CS=0;PS=83;ES=111;OS=19920;SP=0/0;SO=0;QS=-;PR=157;ER=166;OR=25120;CR=0;SR=0;QR=-;PL=0,0;BL=0;LS=0;RB=0/0;SB=-/-;EN=PCMU;DE=PCMU;JI=31,0;DL=0,0,0;IP=192.168.140.29:7090,192.168.140.29:17412
X-RTP-Stat-Add: DQ=3;DSS=0;DS=0;PLCS=64;JS=0
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
Really destroying SIP dialog '[email protected]' Method: BYE

<--- SIP read from UDP:192.168.140.21:5060 --->

<------------->
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
Really destroying SIP dialog '[email protected]' Method: BYE

<--- SIP read from UDP:192.168.140.21:5060 --->

<------------->

<--- SIP read from UDP:192.168.140.21:5060 --->

<------------->
192*CLI>


traces B nach einem reboot (u.a. Unable to create channel of type 'SIP' (cause 20 - Subscriber absent))
Code: 
192.168.140.29 login: root
Password:
   __  _   __  __ ___ __
  |__ |_) |__ |__  |   /
  |   |\  |__ |__  |  /_

   The fun has just begun ...


BusyBox v1.22.1 (2014-02-04 21:41:53 CET) built-in shell (ash)
Enter 'help' for a list of built-in commands.

root@192:/var/mod/root# asterisk -vvvvvc
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk -r' to connect.
root@192:/var/mod/root# asterisk -rvvv
Asterisk 11.8.0-rc1, Copyright (C) 1999 - 2013 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 11.8.0-rc1 currently running on 192 (pid = 2010)
    -- Remote UNIX connection
192*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:192.168.140.21:5060 --->

<------------->

<--- SIP read from UDP:192.168.140.21:5060 --->

<------------->

<--- SIP read from UDP:192.168.140.29:5060 --->
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK6A07E4BC1A6AF254
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=08BC68F61DCB317F
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
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: 423

v=0
o=user 12569997 12569997 IN IP4 192.168.140.29
s=call
c=IN IP4 192.168.140.29
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 192.168.140.29:5060 (NAT)
Sending to 192.168.140.29:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '620' for '***124#*#**610' from 192.168.140.29:5060
  == Using SIP RTP CoS mark 5
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 - (ulaw|alaw), peer - audio=(ulaw|alaw|g726|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.29:7082
Looking for 621 in fritzbox (domain 192.168.140.29)
list_route: hop: <sip:[email protected]>

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK6A07E4BC1A6AF254;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=08BC68F61DCB317F
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [621@fritzbox:1] Dial("SIP/620-00000000", "SIP/2001") in new stack
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
[Feb 13 20:20:07] WARNING[3711][C-00000000]: app_dial.c:2438 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/620-00000000' status is 'CHANUNAVAIL'

<--- Reliably Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK6A07E4BC1A6AF254;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=08BC68F61DCB317F
To: <sip:[email protected]:5061>;tag=as0390b134
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.140.29:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK6A07E4BC1A6AF254
From: "Mobilteil 1" <sip:***124%23*%23**[email protected]>;tag=08BC68F61DCB317F
To: <sip:[email protected]:5061>;tag=as0390b134
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:192.168.140.29:5060 --->
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK4E1391C7D7B58596
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=15F28B4D3DAC3CF9
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
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: 421

v=0
o=user 7230592 7230592 IN IP4 192.168.140.29
s=call
c=IN IP4 192.168.140.29
t=0 0
m=audio 7078 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:7079
<------------->
--- (17 headers 18 lines) ---
Sending to 192.168.140.29:5060 (NAT)
Sending to 192.168.140.29:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '620' for '***123#*#**610' from 192.168.140.29:5060
  == Using SIP RTP CoS mark 5
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 - (ulaw|alaw), peer - audio=(ulaw|alaw|g726|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.29:7078
Looking for 620 in fritzbox (domain 192.168.140.29)
list_route: hop: <sip:[email protected]>

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK4E1391C7D7B58596;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=15F28B4D3DAC3CF9
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [620@fritzbox:1] Dial("SIP/620-00000001", "SIP/2000") in new stack
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
[Feb 13 20:20:21] WARNING[3712][C-00000001]: app_dial.c:2438 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/620-00000001' status is 'CHANUNAVAIL'

<--- Reliably Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK4E1391C7D7B58596;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=15F28B4D3DAC3CF9
To: <sip:[email protected]:5061>;tag=as56c7ba45
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.140.29:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK4E1391C7D7B58596
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=15F28B4D3DAC3CF9
To: <sip:[email protected]:5061>;tag=as56c7ba45
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:192.168.140.21:5060 --->

<------------->

<--- SIP read from UDP:192.168.140.21:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK005c96f95993e311ab8c08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=645881951
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 61 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:[email protected]>
Content-Length: 418

v=0
o=- 1926885678 0 IN IP4 192.168.140.41
s=SIPPER for PhonerLite
c=IN IP4 192.168.140.41
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:4130425362
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Sending to 192.168.140.21:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '2000' for '2000' from 192.168.140.21:5060

<--- Reliably Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK005c96f95993e311ab8c08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=645881951
To: <sip:[email protected]>;tag=as629492ef
Call-ID: [email protected]
CSeq: 61 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="038a1c00"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.140.21:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK005c96f95993e311ab8c08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=645881951
To: <sip:[email protected]>;tag=as629492ef
Call-ID: [email protected]
CSeq: 61 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.140.21:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK005c96f95993e311ab8d08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=645881951
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 62 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="038a1c00", uri="sip:[email protected]", response="db219638c7127005f1610fa0188884b2", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:[email protected]>
Content-Length: 418

v=0
o=- 1926885678 0 IN IP4 192.168.140.21
s=SIPPER for PhonerLite
c=IN IP4 192.168.140.21
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:4130425362
a=sendrecv
<------------->
--- (15 headers 18 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '2000' for '2000' from 192.168.140.21:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|g726|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.21:5062
Looking for 2001 in localphone (domain 192.168.140.29)
list_route: hop: <sip:[email protected]:5060>

<--- Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK005c96f95993e311ab8d08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=645881951
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 62 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [2001@localphone:1] Dial("SIP/2000-00000002", "SIP/2001") in new stack
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
[Feb 13 20:20:42] WARNING[3715][C-00000002]: app_dial.c:2438 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [2001@localphone:2] VoiceMail("SIP/2000-00000002", "2001,u") in new stack
Audio is at 16644
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK005c96f95993e311ab8d08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=645881951
To: <sip:[email protected]>;tag=as2aa7767a
Call-ID: [email protected]
CSeq: 62 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5061>
Content-Type: application/sdp
Content-Length: 294

v=0
o=root 1218930689 1218930689 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 16644 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 - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:192.168.140.21:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK005c96f95993e311ab8e08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=645881951
To: <sip:[email protected]>;tag=as2aa7767a
Call-ID: [email protected]
CSeq: 62 ACK
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="038a1c00", uri="sip:[email protected]:5061", response="d31271118f060c4e4592b5901a083173", algorithm=MD5
Max-Forwards: 70
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
[Feb 13 20:20:42] WARNING[3715][C-00000002]: app_voicemail.c:6322 leave_voicemail: No entry in voicemail config file for '2001'
    -- Auto fallthrough, channel 'SIP/2000-00000002' status is 'CHANUNAVAIL'

<--- SIP read from UDP:192.168.140.21:5060 --->
BYE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK801f60fb5993e311ab8e08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=645881951
To: <sip:[email protected]>;tag=as2aa7767a
Call-ID: [email protected]
CSeq: 63 BYE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="038a1c00", uri="sip:[email protected]:5061", response="27d866833fe2698e4c0c7af14f6f0d9f", algorithm=MD5
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK801f60fb5993e311ab8e08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=645881951
To: <sip:[email protected]>;tag=as2aa7767a
Call-ID: [email protected]
CSeq: 63 BYE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.140.29:5060 --->
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK3985EC5D81E8AD60
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=52614CD2433F06E8
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
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: 421

v=0
o=user 8088771 8088771 IN IP4 192.168.140.29
s=call
c=IN IP4 192.168.140.29
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 192.168.140.29:5060 (NAT)
Sending to 192.168.140.29:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '620' for '***123#*#**610' from 192.168.140.29:5060
  == Using SIP RTP CoS mark 5
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 - (ulaw|alaw), peer - audio=(ulaw|alaw|g726|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.29:7082
Looking for 620 in fritzbox (domain 192.168.140.29)
list_route: hop: <sip:[email protected]>

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK3985EC5D81E8AD60;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=52614CD2433F06E8
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [620@fritzbox:1] Dial("SIP/620-00000003", "SIP/2000") in new stack
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
[Feb 13 20:20:51] WARNING[3718][C-00000003]: app_dial.c:2438 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/620-00000003' status is 'CHANUNAVAIL'

<--- Reliably Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK3985EC5D81E8AD60;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=52614CD2433F06E8
To: <sip:[email protected]:5061>;tag=as66be9d71
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.140.29:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK3985EC5D81E8AD60
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=52614CD2433F06E8
To: <sip:[email protected]:5061>;tag=as66be9d71
Call-ID: [email protected]
CSeq: 2 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:192.168.140.21:5060 --->

<------------->

<--- SIP read from UDP:192.168.140.21:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK002ca90c5a93e311ab8f08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=1448527995
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 64 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:[email protected]>
Content-Length: 418

v=0
o=- 2785899141 0 IN IP4 192.168.140.41
s=SIPPER for PhonerLite
c=IN IP4 192.168.140.41
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3873168774
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Sending to 192.168.140.21:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '2000' for '2000' from 192.168.140.21:5060

<--- Reliably Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK002ca90c5a93e311ab8f08002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=1448527995
To: <sip:[email protected]>;tag=as02d6115f
Call-ID: [email protected]
CSeq: 64 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f32fd53"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.140.21:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK002ca90c5a93e311ab8f08002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=1448527995
To: <sip:[email protected]>;tag=as02d6115f
Call-ID: [email protected]
CSeq: 64 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.140.21:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK002ca90c5a93e311ab9008002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=1448527995
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 65 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="3f32fd53", uri="sip:[email protected]", response="7b70ea7ef59fc5b7335cb5f3b006e7af", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:[email protected]>
Content-Length: 418

v=0
o=- 2785899141 0 IN IP4 192.168.140.21
s=SIPPER for PhonerLite
c=IN IP4 192.168.140.21
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3873168774
a=sendrecv
<------------->
--- (15 headers 18 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '2000' for '2000' from 192.168.140.21:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|g726|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.21:5062
Looking for 2001 in localphone (domain 192.168.140.29)
list_route: hop: <sip:[email protected]:5060>

<--- Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK002ca90c5a93e311ab9008002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=1448527995
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 65 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [2001@localphone:1] Dial("SIP/2000-00000004", "SIP/2001") in new stack
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
[Feb 13 20:21:14] WARNING[3722][C-00000004]: app_dial.c:2438 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [2001@localphone:2] VoiceMail("SIP/2000-00000004", "2001,u") in new stack
Audio is at 10524
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK002ca90c5a93e311ab9008002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=1448527995
To: <sip:[email protected]>;tag=as49ec76b2
Call-ID: [email protected]
CSeq: 65 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5061>
Content-Type: application/sdp
Content-Length: 294

v=0
o=root 1798897403 1798897403 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 10524 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 - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:192.168.140.21:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK80c2410d5a93e311ab9008002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=1448527995
To: <sip:[email protected]>;tag=as49ec76b2
Call-ID: [email protected]
CSeq: 65 ACK
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="3f32fd53", uri="sip:[email protected]:5061", response="06c7f2b5d5a3c3733a32bb4973e6a71f", algorithm=MD5
Max-Forwards: 70
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
[Feb 13 20:21:14] WARNING[3722][C-00000004]: app_voicemail.c:6322 leave_voicemail: No entry in voicemail config file for '2001'
    -- Auto fallthrough, channel 'SIP/2000-00000004' status is 'CHANUNAVAIL'
Really destroying SIP dialog '[email protected]' Method: BYE
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.140.21:5060
Reliably Transmitting (NAT) to 192.168.140.21:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK50dc7d9a;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as49ec76b2
To: "2000" <sip:[email protected]>;tag=1448527995
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.8.0-rc1
Proxy-Authorization: Digest username="2000", realm="asterisk", algorithm=MD5, uri="sip:192.168.140.29", nonce="3f32fd53", response="8073c28ca1b5efe1d5722cb34c510c1b"
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0


---

<--- SIP read from UDP:192.168.140.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.29:5061;branch=z9hG4bK50dc7d9a;rport=5061
From: <sip:[email protected]>;tag=as49ec76b2
To: "2000" <sip:[email protected]>;tag=1448527995
Call-ID: [email protected]
CSeq: 102 BYE
Contact: <sip:[email protected]:5060>
Server: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:192.168.140.21:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK80de8f1b5a93e311ab9108002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=647467175
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 66 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:[email protected]>
Content-Length: 418

v=0
o=- 2876076235 0 IN IP4 192.168.140.41
s=SIPPER for PhonerLite
c=IN IP4 192.168.140.41
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3532576708
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Sending to 192.168.140.21:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '2000' for '2000' from 192.168.140.21:5060

<--- Reliably Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK80de8f1b5a93e311ab9108002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=647467175
To: <sip:[email protected]>;tag=as13fc5e1a
Call-ID: [email protected]
CSeq: 66 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="630a3acc"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.140.21:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.41:5060;branch=z9hG4bK80de8f1b5a93e311ab9108002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=647467175
To: <sip:[email protected]>;tag=as13fc5e1a
Call-ID: [email protected]
CSeq: 66 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.140.21:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK80de8f1b5a93e311ab9208002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=647467175
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 67 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="630a3acc", uri="sip:[email protected]", response="3fe46857debdcc363f67e7445e381cf2", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:[email protected]>
Content-Length: 418

v=0
o=- 2876076235 0 IN IP4 192.168.140.21
s=SIPPER for PhonerLite
c=IN IP4 192.168.140.21
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3532576708
a=sendrecv
<------------->
--- (15 headers 18 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '2000' for '2000' from 192.168.140.21:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|g726|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.21:5062
Looking for 2001 in localphone (domain 192.168.140.29)
list_route: hop: <sip:[email protected]:5060>

<--- Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK80de8f1b5a93e311ab9208002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=647467175
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 67 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [2001@localphone:1] Dial("SIP/2000-00000005", "SIP/2001") in new stack
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
[Feb 13 20:21:39] WARNING[3723][C-00000005]: app_dial.c:2438 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [2001@localphone:2] VoiceMail("SIP/2000-00000005", "2001,u") in new stack
Audio is at 11636
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK80de8f1b5a93e311ab9208002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=647467175
To: <sip:[email protected]>;tag=as71764dbb
Call-ID: [email protected]
CSeq: 67 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5061>
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 144575753 144575753 IN IP4 192.168.140.29
s=Asterisk PBX 11.8.0-rc1
c=IN IP4 192.168.140.29
t=0 0
m=audio 11636 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 - - - -
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:192.168.140.21:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK80de8f1b5a93e311ab9308002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=647467175
To: <sip:[email protected]>;tag=as71764dbb
Call-ID: [email protected]
CSeq: 67 ACK
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="630a3acc", uri="sip:[email protected]:5061", response="d0a73d386fa5159de3e00e58965297b4", algorithm=MD5
Max-Forwards: 70
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
[Feb 13 20:21:39] WARNING[3723][C-00000005]: app_voicemail.c:6322 leave_voicemail: No entry in voicemail config file for '2001'
    -- Auto fallthrough, channel 'SIP/2000-00000005' status is 'CHANUNAVAIL'

<--- SIP read from UDP:192.168.140.21:5060 --->
BYE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK809254205a93e311ab9308002700ec4a;rport
From: "2000" <sip:[email protected]>;tag=647467175
To: <sip:[email protected]>;tag=as71764dbb
Call-ID: [email protected]
CSeq: 68 BYE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="630a3acc", uri="sip:[email protected]:5061", response="74b313cb96c8128cbf1cb42ee359a758", algorithm=MD5
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.140.21:5060 (NAT)
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.140.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.21:5060;branch=z9hG4bK809254205a93e311ab9308002700ec4a;received=192.168.140.21;rport=5060
From: "2000" <sip:[email protected]>;tag=647467175
To: <sip:[email protected]>;tag=as71764dbb
Call-ID: [email protected]
CSeq: 68 BYE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '[email protected]' Method: BYE

<--- SIP read from UDP:192.168.140.29:5060 --->
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK6A26193D654D8DFE
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=9DC6CE206278FF02
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 3 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
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: 421

v=0
o=user 7764165 7764165 IN IP4 192.168.140.29
s=call
c=IN IP4 192.168.140.29
t=0 0
m=audio 7078 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:7079
<------------->
--- (17 headers 18 lines) ---
Sending to 192.168.140.29:5060 (NAT)
Sending to 192.168.140.29:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '620' for '***123#*#**610' from 192.168.140.29:5060
  == Using SIP RTP CoS mark 5
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 - (ulaw|alaw), peer - audio=(ulaw|alaw|g726|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.140.29:7078
Looking for 620 in fritzbox (domain 192.168.140.29)
list_route: hop: <sip:[email protected]>

<--- Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK6A26193D654D8DFE;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=9DC6CE206278FF02
To: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 3 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5061>
Content-Length: 0


<------------>
    -- Executing [620@fritzbox:1] Dial("SIP/620-00000006", "SIP/2000") in new stack
Really destroying SIP dialog '[email protected]:5061' Method: INVITE
[Feb 13 20:21:50] WARNING[3724][C-00000006]: app_dial.c:2438 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/620-00000006' status is 'CHANUNAVAIL'

<--- Reliably Transmitting (NAT) to 192.168.140.29:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK6A26193D654D8DFE;received=192.168.140.29;rport=5060
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=9DC6CE206278FF02
To: <sip:[email protected]:5061>;tag=as40e6b411
Call-ID: [email protected]
CSeq: 3 INVITE
Server: Asterisk PBX 11.8.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.140.29:5060 --->
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.140.29:5060;branch=z9hG4bK6A26193D654D8DFE
From: "Mobilteil 1" <sip:***123%23*%23**[email protected]>;tag=9DC6CE206278FF02
To: <sip:[email protected]:5061>;tag=as40e6b411
Call-ID: [email protected]
CSeq: 3 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.03 (Feb 7 2014)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK
192*CLI>
 
Zuletzt bearbeitet:
Das Problem scheint wie angenommen ein Timing Problem zu sein.

Problem: Nach einem Reboot kann ich weder von extern noch von PhonerLite die Rufnummer 2000 auf der Fritzbox (fb7390), auf der Asterisk läuft erreichen.
Eine schlechte Lösung: Wenn ich jetzt die Rufnummer auf der Fritzbox deaktiviere und anschließend wieder aktiviere, dann kann ich eine Verbindung zu der Rufnummer 2000 erreichen.
Nicht viel besser: Ich habe meine Konfiguration auf eine fb7270v3 kopiert. Da läuft alles zur Zeit einwandfrei :) -- svn tag 11757 mit 05.54

Hat jemand bereits ein DECT Telefon (oder intigrierten AB) auf einer fb7390 auf der Asterisk läuft mit dem aktuellen Asterisk (11.8.0-rc1) ans laufen gebracht?
Welche Einstellungen waren dafür notwendig?
Hat jemand das ganze schon auf einer fb7490 probiert (da möchte ich Leistungsbedingt hin)?

Vielen Dank für eure Tips.
/Andreas
 
Zuletzt bearbeitet:
Mit dem neuen Asterisk 11.8.0-rc2 tritt das Problem nicht mehr auf.
Anpassen musste ich das AVM Forewarding.
Zur Zeit habe ich forewards für 5060+101 und 10000+10001 jeweils für TCP und UDP eingerichtet.

Wichtig ist, dass das port forwarding auch den Port 5060 eischließt.
 
Zuletzt bearbeitet:
Hallo,

Das Problem mit "re-open master file" hatte ich wie folgt gelöst:
- Anlegen eines Verzeichnisses "asterisk.log" mit Unterverzeichnis "cdr-csv" auf dem USB stick.
- Verändern des Log pfades in der asterisk.conf auf "/var/media/ftp/uStor01/asterisk.log"

Damit war das Problem mit der Fehlermeldung gelöst.
Allerdings kann es dann irgendwann ein Problem geben, wenn die Log Datei zu groß wird!

/Andreas
 
Tja, manchmal sieht man den Wald vor lauter Bäumen nicht.

Danke für die Demonstration des Einfachen [Denkens]!;)

Hat geklappt!
 
Timingproblem

andreas96k schrieb:
Das Problem scheint wie angenommen ein Timing Problem zu sein...
Zum Vergrößern anklicken....

Das Timing-Problem mit unkalkulierbaren Verhalten nach jedem Bootvorgang, welches u.a. externe SIP-Anrufe mit
"Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1)"
verhinderte - bis zu einem manuellen Asterisk-Neustart ohne Reboot, konnte ich unter opensuse 13.1 lösen damit, Asterisk beim Booten erst NACH Netzwerkinitialisierung zu starten.

in
/etc/init.d/systemd/system/basic.target.wants/asterisk.service:

Code: 
[Unit]
Description=Asterisk PBX
After=network.target
Wants=nss-lookup.target

[Service]
PIDFile=/run/asterisk/asterisk.pid
ExecStart=/usr/sbin/asterisk -fn
ExecReload=/bin/kill -HUP $MAINPID

[Install]
WantedBy=basic.target

Gruß
Walter
 
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