SIP-Einstellung für Port wird ignoriert (port 5060 eingestellt, aber Port 36112 wird

apfelmaennchen · 6 Mai 2013

Ich habe in sip.conf in [general] port=5060 und in dem Teil für meinen sip-client auch port=5060.

Trotzdem erscheint im CLI bei Kommando "sip show peers": (IP teils ausgex'ed)

Name/username Host Dyn Forcerport ACL Port
SIP_CLIENT1/SIP_CLIENT1 109.xx.x.xx D N 36112

Hat jemand eine Idee? Ich habe natürlich den Asterisk neu gestartet.

Hier die Konfiguration der angesprochenen SIP-clients in der SIP.conf:

Code:

[SIP_CLIENT1] 
type=friend
transport=tcp
nat=yes      
qualify=yes  ;force keepalives
port=5060
context=sipgate 
secret=meinpassword
host=dynamic
;encryption=yes ; SRTP Encryption
registertimeout = 600
disallow=all
dtmfmode=rfc2833
insecure=invite
allow=alaw
allow=ilbc
allow=g723
allow=g722
allow=siren7
allow=siren14
allow=g726
allow=g729

rollo · 6 Mai 2013

Entweder Du hast den Port im Router nicht geforwardet oder, was ich anhand der IP eher vermute, du bist hinter Provider NAT.
Im zweiten Fall hast du keince Chance den Asterisk von aussen zu erreichen. Eine Verbindung nach extern sollte aber kein Problem sein.

jo

MaverrickTM · 6 Mai 2013

Führe mal bitte das folgende auf einer Shell aus und Poste das Ergebnis:

Code:

netstat -nlp

[EDIT]
Im übrigen wäre es hilfreich zu erfahren welche Asterisk Version Du benutzt, da sich mitunter die Parameter geändert haben.
[/EDIT]

apfelmaennchen · 6 Mai 2013

Aktive Internetverbindungen (Nur Server)

Code:

Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
tcp        0      0 0.0.0.0:5060            0.0.0.0:*               LISTEN      1579/asterisk   
tcp        0      0 127.0.0.1:3306          0.0.0.0:*               LISTEN      1239/mysqld     
tcp        0      0 0.0.0.0:110             0.0.0.0:*               LISTEN      1350/cyrmaster  
tcp        0      0 0.0.0.0:143             0.0.0.0:*               LISTEN      1350/cyrmaster  
tcp        0      0 0.0.0.0:10000           0.0.0.0:*               LISTEN      1687/perl       
tcp        0      0 127.0.0.1:2000          0.0.0.0:*               LISTEN      1350/cyrmaster  
tcp        0      0 0.0.0.0:53              0.0.0.0:*               LISTEN      1151/dnsmasq    
tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      1162/sshd       
tcp        0      0 0.0.0.0:119             0.0.0.0:*               LISTEN      1350/cyrmaster  
tcp        0      0 0.0.0.0:25              0.0.0.0:*               LISTEN      1532/master     
tcp6       0      0 127.0.0.1:8005          :::*                    LISTEN      1461/java       
tcp6       0      0 :::110                  :::*                    LISTEN      1350/cyrmaster  
tcp6       0      0 :::143                  :::*                    LISTEN      1350/cyrmaster  
tcp6       0      0 :::8080                 :::*                    LISTEN      1461/java       
tcp6       0      0 :::80                   :::*                    LISTEN      1638/apache2    
tcp6       0      0 :::21                   :::*                    LISTEN      1595/proftpd: (acce
tcp6       0      0 :::53                   :::*                    LISTEN      1151/dnsmasq    
tcp6       0      0 :::22                   :::*                    LISTEN      1162/sshd       
tcp6       0      0 :::119                  :::*                    LISTEN      1350/cyrmaster  
udp        0      0 0.0.0.0:10000           0.0.0.0:*                           1687/perl       
udp        0      0 0.0.0.0:53              0.0.0.0:*                           1151/dnsmasq    
udp        0      0 0.0.0.0:5060            0.0.0.0:*                           1579/asterisk   
udp        0      0 0.0.0.0:69              0.0.0.0:*                           1414/inetutils-inet
udp        0      0 0.0.0.0:4569            0.0.0.0:*                           1579/asterisk   
udp6       0      0 :::53                   :::*                                1151/dnsmasq

Aktive Sockets in der UNIX-Dom?ne (Nur Server)

Code:

Proto RefCnt Flags       Type       State         I-Node   PID/Program name    Pfad
unix  2      [ ACC ]     STREAM     H?RT          4251     1532/master         private/local
unix  2      [ ACC ]     STREAM     H?RT          4172     1532/master         public/cleanup
unix  2      [ ACC ]     STREAM     H?RT          4179     1532/master         private/tlsmgr
unix  2      [ ACC ]     STREAM     H?RT          4183     1532/master         private/rewrite
unix  2      [ ACC ]     STREAM     H?RT          4187     1532/master         private/bounce
unix  2      [ ACC ]     STREAM     H?RT          4255     1532/master         private/virtual
unix  2      [ ACC ]     STREAM     H?RT          4259     1532/master         private/lmtp
unix  2      [ ACC ]     STREAM     H?RT          3968     1350/cyrmaster      /var/run/cyrus/socket/lmtp
unix  2      [ ACC ]     STREAM     H?RT          4263     1532/master         private/anvil
unix  2      [ ACC ]     STREAM     H?RT          4267     1532/master         private/scache
unix  2      [ ACC ]     STREAM     H?RT          3206     1136/dbus-daemon    /var/run/dbus/system_bus_socket
unix  2      [ ACC ]     STREAM     H?RT          4271     1532/master         private/maildrop
unix  2      [ ACC ]     STREAM     H?RT          4275     1532/master         private/uucp
unix  2      [ ACC ]     STREAM     H?RT          4279     1532/master         private/ifmail
unix  2      [ ACC ]     STREAM     H?RT          4322     1558/saslauthd      /var/run/saslauthd/mux
unix  2      [ ACC ]     STREAM     H?RT          4191     1532/master         private/defer
unix  2      [ ACC ]     STREAM     H?RT          4363     1579/asterisk       /var/run/asterisk/asterisk.ctl
unix  2      [ ACC ]     STREAM     H?RT          4283     1532/master         private/bsmtp
unix  2      [ ACC ]     STREAM     H?RT          4198     1532/master         private/trace
unix  2      [ ACC ]     STREAM     H?RT          4287     1532/master         private/scalemail-backend
unix  2      [ ACC ]     STREAM     H?RT          4291     1532/master         private/mailman
unix  2      [ ACC ]     STREAM     H?RT          4202     1532/master         private/verify
unix  2      [ ACC ]     STREAM     H?RT          4206     1532/master         public/flush
unix  2      [ ACC ]     STREAM     H?RT          4214     1532/master         private/proxymap
unix  2      [ ACC ]     STREAM     H?RT          4219     1532/master         private/proxywrite
unix  2      [ ACC ]     STREAM     H?RT          4223     1532/master         private/smtp
unix  2      [ ACC ]     STREAM     H?RT          4227     1532/master         private/relay
unix  2      [ ACC ]     STREAM     H?RT          4235     1532/master         public/showq
unix  2      [ ACC ]     STREAM     H?RT          4239     1532/master         private/error
unix  2      [ ACC ]     STREAM     H?RT          3363     1239/mysqld         /var/run/mysqld/mysqld.sock
unix  2      [ ACC ]     STREAM     H?RT          4243     1532/master         private/retry
unix  2      [ ACC ]     STREAM     H?RT          4247     1532/master         private/discard

MaverrickTM · 6 Mai 2013

Dein Asterisk lauscht nur auf den folgenden Ports:

- 5060 (TCP & UDP)
- 4569 (UDP)

Daher ist alles gut.

apfelmaennchen · 6 Mai 2013

Beitrag 1
Ich nutze einen V-Server und bin hinter keinem NAT oder Router. Generell erreiche ich Asterisk, meist sogar qualitätiv gut. Nur habe ich manchmal gar kein Audio oder nur one-way.

Beitrag 2
Connected to Asterisk 1.8.11-cert2

Beitrag 3

Hi Maverrick, danke!
JA, ich habe iptables so konfiguriert, dass es nur auf 5060 lauscht. Aber dennoch wundert mich dass bei "SIP SHOW PEERS" es so aussieht, als ob mein IPHONE_GUS auf einem anderen PORT angemeldet ist. Siehe screenie:

rollo · 6 Mai 2013

Die (versteckte) Frage nach der Netzanbindung ist noch unbeantwortet.

Bitte keine Beiträge direkt nacheinander. Benutze den "Bearbeiten"-Button wenn Die noch etwas einfällt.

jo

MaverrickTM · 6 Mai 2013

Eventuell wird hier nicht der Zielport angezeigt sondern der Quellport des SIP-Devices. Hab ab und an ebenfalls solch einen "ausreisser". Das kannst Du wunderbar mit tcpdump überprüfen!

Code:

tcpdump -nnni eth0 host <hier die IP des SIP-Gerätes>

Bzgl. Deiner sporadischen Probleme: Das ist etwas merkwürdig, weils ja tendenziell funktioniert - nur halt ab und an nicht. werden irgendwie automatisiert Änderungen(e.g. Fail2Ban o.ä.) an dem iptables Regelwerk vorgenommen?

[EDIT]
Hab meinen "ausreisser gefunden":

Code:

204/204                    10.0.0.2                                 D   N             20066    OK (161 ms)

tcpdump zeigt folgendes:

Code:

12:59:11.811186 IP 10.0.0.2.20066 > 10.0.0.105.5060: SIP, length: 4

Es ist also wie vermutet. Das was dort angezeigt wird, ist nicht der lauschende Port des Asterisk - sondern der Quellport des SIP-Clients.
[/EDIT]

apfelmaennchen · 6 Mai 2013

Super. Vielen Dank, Mav.

Nein, ich habe die iptables fix eingerichtet, die werden nicht dynamisch angepasst.

rentier-s · 6 Mai 2013

Ich versteh die Aufregung hier nicht ganz. sip show peers zeigt an, auf welchem Port das Endgerät zu erreichen ist. Da kann Asterisk nichts dran machen. Das ist ggf. lediglich für ein Portforwarding auf der Client-Seite interessant.

Die Angabe von port im general bringt eigentlich gar nichts. Sofern Asterisk nicht auf 5060 erreichbar sein soll, ist hier bindport das Richtige. Ebenso ist die Angabe von port bei Clients nutzlos, die sich am Asterisk registrieren (i.V.m. host=dynamic), wie man eingangs ja schön sieht.

Mit dem SIP Port haben Audio-Probleme sowieso nichts zu tun. Dafür ist der RTP Portbereich da. Wenn sporadisch kein RTP Strom durch kommt, müsste man das mittels SIP Debug beobachten, ob vom Client evtl. falsche weil ge-nat-ete Ports/Adressen angegeben werden.

apfelmaennchen · 6 Mai 2013

#1 Hi Rentier. Auch danke. Ich will NICHT nicht 5060 benutzen, sondern ich will genau NUR 5060 verwenden. Nämlich deshalb, damit ich dafür eine exklusive Regel in Iptables einrichten kann.

Wenn ich beim CLIENT als auch beim SERVER keine NAT davor habe, ist es dann ratsam, nat=no einzustellen?

#2 Provider hat bestätigt: keine Firewall oder NAT oder geblockter PORT vor meiner Maschine.

rentier-s · 7 Mai 2013

Wie gesagt, den Port des Clients kannst Du Asterisk-seitig nicht beeinflussen.

In Iptables brauchst Du halt ankommend UDP und TCP mit Zielport 5060, und ankommend UDP mit Zielportbereich aus der rtp.conf. Wenn Du abgehend einschränken willst, dann mit den o.g. Ports als Quellports.

Sofern tatsächlich kein NAT beteiligt ist, kannst Du nat einfach weglassen. Du schreibst aber was von Iphone, also versuchst Du vermutlich über UMTS zu VoIPen, oder? Dann wärst Du nämlich ziemlich sicher hinter einer NAT.

Und wie auch bereits gesagt, Audio-Probleme bitte mit sip set debug on analysieren.

apfelmaennchen · 7 Mai 2013

Danke! (gibt keinen Danke-button hier im Forum, gell?)

Dein Hinweis

Sofern tatsächlich kein NAT beteiligt ist, kannst Du nat einfach weglassen. Du schreibst aber was von Iphone, also versuchst Du vermutlich über UMTS zu VoIPen, oder? Dann wärst Du nämlich ziemlich sicher hinter einer NAT.

ist noch sehr hilfreich. Du hast Recht, in der Tat baue ich mit iphone im UMTS-Netz von VODAFONE (auch teilweise SWISSCOM) die Datenverbindung auf. Bin ich da hinter einer NAT? Kann man das beeinflussen oder kann man damit das iphone als SIP-client vergessen? Seit mehr als einem Jahr nun bekomme ich trotz aller Fummelei und diversen CLIENTS die NAT-traversial issues nicht in den Griff. Am stabilsten wars noch mit groundwire und einer "NAT BRIDGE".

Hast Du einen Tipp, ob/wie ich überhaupt herausfinde, ob ich mit dem iphone hinter einem NAT bin und wie ich das traversal issue lösen kann?

rentier-s · 7 Mai 2013

Da ich kein Iphone habe, weiß ich nicht wie man da seine IP-Adresse auslesen kann. Jedenfalls könntest Du diese mit dem Ergebnis einer der vielen Internetseiten vergleichen, die die eigene Adresse anzeigen. Unterscheiden sie sich, ist ein Router dazwischen.

Setze in der sip.conf für den/die Client(s) nat=yes (oder entsprechendes in Asterisk Versionen [noparse]>1.8)[/noparse] und directmedia=no. Praktisch jeder SIP Client bietet ICE und/oder STUN, als Stun-Server kannst Du jeden beliebigen nutzen. Außerdem kann es helfen, wenn der Client selbst keepalive bietet, damit die Route offen bleibt.

Für alles weitere (zum 3. Mal) sip set debug on, dann siehst Du genau welche Adressen für den Datenstrom angegeben werden.

apfelmaennchen · 10 Mai 2013

Hi Rentier, ich habe "sip set debug on" und würde Dir gerne ein SIP-log als PM schicken, die ich gerade aufgezeichnet habe von einem Call der one-way Audio hatte.

MaverrickTM · 10 Mai 2013

Du kannst Die Ausgabe auch hier posten. Packe die Ausgabe aber bitte in Code-Tags (Der # Button im Erweiterten Modus - oder [ code ] Hier Deine Ausgabe [ /code ] (ohne Leerzeichen in den eckigen Klammern))!

apfelmaennchen · 10 Mai 2013

Code:

gref03*CLI> !date
Fr 10. Mai 11:38:35 CEST 2013
  == Using SIP RTP CoS mark 5
    -- Executing [200@sipgate:1] Verbose("SIP/GUS_FRITZ-000009c4", "call to GUS_iphone") in new stack
call to GUS_iphone
    -- Executing [200@sipgate:2] Set("SIP/GUS_FRITZ-000009c4", "CALLEDNUMBER="200"") in new stack
    -- Executing [200@sipgate:3] Dial("SIP/GUS_FRITZ-000009c4", "SIP/GUS_iphone") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/GUS_iphone
    -- SIP/GUS_iphone-000009c5 is ringing
gref03*CLI> sip set debug on
SIP Debugging enabled
Really destroying SIP dialog '130F84392CE80D8393E52C71B7D47D18B8588732' Method: REGISTER

<--- SIP read from TCP:92.50.92.74:61378 --->
REGISTER sip:212.40.173.65 SIP/2.0
Via: SIP/2.0/TCP 92.50.92.74:61378;branch=z9hG4bKdmZQZY3LAnramd2j;rport
Contact: <sip:[email protected]:61378;rinstance=02A6C622;transport=tcp>;expires=600
Max-Forwards: 70
From: "sipgate_trunk" <sip:[email protected]>;tag=4546F46E2CDBA405B759C8425F1C98AD
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.9
To: "sipgate_trunk" <sip:[email protected]>
Expires: 600
Call-ID: 105FFECF567D0FB80CC4905CC8CFA0EF1A8BDAF0
CSeq: 35129 REGISTER
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 92.50.92.74:61378 (NAT)

<--- Transmitting (NAT) to 92.50.92.74:61378 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 92.50.92.74:61378;branch=z9hG4bKdmZQZY3LAnramd2j;received=92.50.92.74;rport=61378
From: "sipgate_trunk" <sip:[email protected]>;tag=4546F46E2CDBA405B759C8425F1C98AD
To: "sipgate_trunk" <sip:[email protected]>;tag=as7038ea4b
Call-ID: 105FFECF567D0FB80CC4905CC8CFA0EF1A8BDAF0
CSeq: 35129 REGISTER
Server: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30fc3935"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '105FFECF567D0FB80CC4905CC8CFA0EF1A8BDAF0' in 32000 ms (Method: REGISTER)

<--- SIP read from TCP:92.50.92.74:61378 --->
REGISTER sip:212.40.173.65 SIP/2.0
Via: SIP/2.0/TCP 92.50.92.74:61378;branch=z9hG4bKj8BZDipwg1bNzGez;rport
Contact: <sip:[email protected]:61378;rinstance=02A6C622;transport=tcp>;expires=600
Max-Forwards: 70
From: "sipgate_trunk" <sip:[email protected]>;tag=4546F46E2CDBA405B759C8425F1C98AD
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.9
To: "sipgate_trunk" <sip:[email protected]>
Expires: 600
Call-ID: 105FFECF567D0FB80CC4905CC8CFA0EF1A8BDAF0
CSeq: 35130 REGISTER
Authorization: Digest username="GUS_iphone",realm="asterisk",algorithm=MD5,uri="sip:212.40.173.65",nonce="30fc3935",response="4c00f3594259894e4235c5a4b6f76cad"
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 92.50.92.74:61378 (NAT)
    -- Registered SIP 'GUS_iphone' at 92.50.92.74:61378
Reliably Transmitting (NAT) to 92.50.92.74:61378:
OPTIONS sip:[email protected]:61378;rinstance=02A6C622;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK6a94fffd;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as55917ad0
To: <sip:[email protected]:61378;rinstance=02A6C622;transport=tcp>
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert2
Date: Fri, 10 May 2013 09:39:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
       > Saved useragent "Acrobits Softphone Business/2.4.9" for peer GUS_iphone

<--- Transmitting (NAT) to 92.50.92.74:61378 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 92.50.92.74:61378;branch=z9hG4bKj8BZDipwg1bNzGez;received=92.50.92.74;rport=61378
From: "sipgate_trunk" <sip:[email protected]>;tag=4546F46E2CDBA405B759C8425F1C98AD
To: "sipgate_trunk" <sip:[email protected]>;tag=as7038ea4b
Call-ID: 105FFECF567D0FB80CC4905CC8CFA0EF1A8BDAF0
CSeq: 35130 REGISTER
Server: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 600
Contact: <sip:[email protected]:61378;rinstance=02A6C622;transport=tcp>;expires=600
Date: Fri, 10 May 2013 09:39:00 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '105FFECF567D0FB80CC4905CC8CFA0EF1A8BDAF0' in 32000 ms (Method: REGISTER)

<--- SIP read from TCP:92.50.92.74:61378 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK6a94fffd;rport=5060;received=212.40.173.65
Contact: <sip:[email protected]:61378;transport=tcp>
From: "asterisk" <sip:[email protected]>;tag=as55917ad0
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
To: <sip:[email protected]:61378;rinstance=02A6C622;transport=tcp>
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
Content-Length: 0

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

<--- SIP read from TCP:92.50.92.74:61378 --->
SUBSCRIBE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 92.50.92.74:61378;branch=z9hG4bKrj7hZreNHSC8jW7G;rport
Contact: <sip:[email protected]:61378;transport=tcp>
Max-Forwards: 70
From: "sipgate_trunk" <sip:[email protected]>;tag=EB218B228EA09F98607C27F44AA03B56
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.9
To: "sipgate_trunk" <sip:[email protected]>
Accept: application/simple-message-summary
Event: message-summary
Expires: 600
Call-ID: 380A7BBD3F7CA2F3DC1A145F8FB3D1458966E900
CSeq: 1 SUBSCRIBE
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Creating new subscription
Sending to 92.50.92.74:61378 (NAT)
list_route: hop: <sip:[email protected]:61378;transport=tcp>
Found peer 'GUS_iphone' for 'GUS_iphone' from 92.50.92.74:61378

<--- Transmitting (NAT) to 92.50.92.74:61378 --->
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/TCP 92.50.92.74:61378;branch=z9hG4bKrj7hZreNHSC8jW7G;received=92.50.92.74;rport=61378
From: "sipgate_trunk" <sip:[email protected]>;tag=EB218B228EA09F98607C27F44AA03B56
To: "sipgate_trunk" <sip:[email protected]>;tag=as70feeeea
Call-ID: 380A7BBD3F7CA2F3DC1A145F8FB3D1458966E900
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[May 10 11:39:00] NOTICE[21815]: chan_sip.c:25201 handle_request_subscribe: Received SIP subscribe for peer without mailbox: GUS_iphone
Really destroying SIP dialog '380A7BBD3F7CA2F3DC1A145F8FB3D1458966E900' Method: SUBSCRIBE
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from TCP:184.73.248.111:55278 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK16e9e7fd;rport=5060;received=212.40.173.65
Contact: <sip:[email protected]:55278;transport=tcp>
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
Call-ID: [email protected]:5060
CSeq: 102 INVITE
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Content-Type: application/sdp
Content-Length: 188

v=0
o=- 50707 11578 IN IP4 192.168.0.11
s=rbtzmef
c=IN IP4 192.168.0.11
t=0 0
m=audio 10416 RTP/AVP 8 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=ptime:10
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format TELEPHONE-EVENT for ID 101
Capabilities: us - 0x7d09 (g723|alaw|g726|g729|ilbc|g722|siren7|siren14), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.11:10416
list_route: hop: <sip:[email protected]:55278;transport=tcp>
set_destination: Parsing <sip:[email protected]:55278;transport=tcp> for address/port to send to
set_destination: set destination to 10.84.66.166:55278
Transmitting (NAT) to 184.73.248.111:55278:
ACK sip:[email protected]:55278;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK32a4d565;rport
Max-Forwards: 70
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.11-cert2
Content-Length: 0


---
    -- SIP/GUS_iphone-000009c5 answered SIP/GUS_FRITZ-000009c4
Audio is at 10006
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 92.50.92.73:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.50.92.73:5060;branch=z9hG4bK7AB7D8C2A1962858;received=92.50.92.73;rport=5060
From: <sip:[email protected]>;tag=FE5449670B25B504
To: <sip:[email protected]>;tag=as346e1e73
Call-ID: [email protected]
CSeq: 3562 INVITE
Server: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 249482761 249482761 IN IP4 212.40.173.65
s=Asterisk PBX 1.8.11-cert2
c=IN IP4 212.40.173.65
t=0 0
m=audio 10006 RTP/AVP 8 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Remotely bridging SIP/GUS_FRITZ-000009c4 and SIP/GUS_iphone-000009c5
set_destination: Parsing <sip:[email protected]:55278;transport=tcp> for address/port to send to
set_destination: set destination to 10.84.66.166:55278
Audio is at 10054
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 184.73.248.111:55278:
INVITE sip:[email protected]:55278;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK3d96016a;rport
Max-Forwards: 70
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2025264092 2025264093 IN IP4 92.50.92.73
s=Asterisk PBX 1.8.11-cert2
c=IN IP4 92.50.92.73
t=0 0
m=audio 7098 RTP/AVP 8 101
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:92.50.92.73:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.50.92.73:5060;branch=z9hG4bK5B1434B3E327947C
From: <sip:[email protected]>;tag=FE5449670B25B504
To: <sip:[email protected]>;tag=as346e1e73
Call-ID: [email protected]
CSeq: 3562 ACK
Contact: <sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4>
Max-Forwards: 70
User-Agent: AVM FRITZ!Box 6360 Cable (um) 85.05.28 (Oct 18 2012)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4> for address/port to send to
set_destination: set destination to 92.50.92.73:5060
Audio is at 10006
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 92.50.92.73:5060:
INVITE sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4 SIP/2.0
Via: SIP/2.0/UDP 212.40.173.65:5060;branch=z9hG4bK5e0aa039;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as346e1e73
To: <sip:[email protected]>;tag=FE5449670B25B504
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 249482761 249482762 IN IP4 92.50.92.74
s=Asterisk PBX 1.8.11-cert2
c=IN IP4 92.50.92.74
t=0 0
m=audio 64005 RTP/AVP 8 101
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:92.50.92.73:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.40.173.65:5060;branch=z9hG4bK5e0aa039;rport=5060
From: <sip:[email protected]>;tag=as346e1e73
To: <sip:[email protected]>;tag=FE5449670B25B504
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4>
User-Agent: AVM FRITZ!Box 6360 Cable (um) 85.05.28 (Oct 18 2012)
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: 245

v=0
o=user 4664846 4664847 IN IP4 92.50.92.73
s=Asterisk PBX 1.8.11-cert2
c=IN IP4 92.50.92.73
t=0 0
m=audio 7098 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:7099
a=ptime:30
<------------->
--- (15 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x1008 (alaw|g722), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 92.50.92.73:7098
set_destination: Parsing <sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4> for address/port to send to
set_destination: set destination to 92.50.92.73:5060
Transmitting (NAT) to 92.50.92.73:5060:
ACK sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4 SIP/2.0
Via: SIP/2.0/UDP 212.40.173.65:5060;branch=z9hG4bK3311bac0;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as346e1e73
To: <sip:[email protected]>;tag=FE5449670B25B504
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.11-cert2
Content-Length: 0


---

<--- SIP read from TCP:184.73.248.111:55278 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK3d96016a;rport=5060;received=212.40.173.65
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
Call-ID: [email protected]:5060
CSeq: 103 INVITE
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Content-Length: 0

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

<--- SIP read from TCP:184.73.248.111:55278 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK3d96016a;rport=5060;received=212.40.173.65
Contact: <sip:[email protected]:55278;transport=tcp>
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
Call-ID: [email protected]:5060
CSeq: 103 INVITE
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Content-Type: application/sdp
Content-Length: 188

v=0
o=- 50707 11578 IN IP4 192.168.0.11
s=rbtzmef
c=IN IP4 192.168.0.11
t=0 0
m=audio 10416 RTP/AVP 8 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=ptime:10
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
set_destination: Parsing <sip:[email protected]:55278;transport=tcp> for address/port to send to
set_destination: set destination to 10.84.66.166:55278
Transmitting (NAT) to 184.73.248.111:55278:
ACK sip:[email protected]:55278;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK0d7c0f9c;rport
Max-Forwards: 70
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.11-cert2
Content-Length: 0


---

<--- SIP read from TCP:184.72.221.84:45357 --->
REGISTER sip:gref03.synserver.de SIP/2.0
Via: SIP/2.0/TCP 10.202.218.207:45357;branch=z9hG4bKVckemm6viqM5RVw1;rport
Contact: <sip:[email protected]:45357;rinstance=0171DB01;transport=tcp>;expires=600
Max-Forwards: 70
From: "alexandra" <sip:[email protected]>;tag=E22ADF38D9CB76615C44E37288997503
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE
Supported: replaces, path
User-Agent: Acrobits SIPIS (http://www.acrobits.cz)
To: "alexandra" <sip:[email protected]>
Expires: 600
Call-ID: F223524B282ECDEC3199E2ECB40B13F1F1270F7D
CSeq: 296 REGISTER
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 184.72.221.84:45357 (NAT)

<--- Transmitting (NAT) to 184.72.221.84:45357 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.202.218.207:45357;branch=z9hG4bKVckemm6viqM5RVw1;received=184.72.221.84;rport=45357
From: "alexandra" <sip:[email protected]>;tag=E22ADF38D9CB76615C44E37288997503
To: "alexandra" <sip:[email protected]>;tag=as5f241421
Call-ID: F223524B282ECDEC3199E2ECB40B13F1F1270F7D
CSeq: 296 REGISTER
Server: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="558faa51"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'F223524B282ECDEC3199E2ECB40B13F1F1270F7D' in 32000 ms (Method: REGISTER)

<--- SIP read from TCP:184.72.221.84:45357 --->
REGISTER sip:gref03.synserver.de SIP/2.0
Via: SIP/2.0/TCP 10.202.218.207:45357;branch=z9hG4bKHGgXr4J3tW3bG8xd;rport
Contact: <sip:[email protected]:45357;rinstance=0171DB01;transport=tcp>;expires=600
Max-Forwards: 70
From: "alexandra" <sip:[email protected]>;tag=E22ADF38D9CB76615C44E37288997503
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE
Supported: replaces, path
User-Agent: Acrobits SIPIS (http://www.acrobits.cz)
To: "alexandra" <sip:[email protected]>
Expires: 600
Call-ID: F223524B282ECDEC3199E2ECB40B13F1F1270F7D
CSeq: 297 REGISTER
Authorization: Digest username="alexandra",realm="asterisk",algorithm=MD5,uri="sip:gref03.synserver.de",nonce="558faa51",response="8839c311edf79703ed59df6dfda3be8b"
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 184.72.221.84:45357 (NAT)
    -- Registered SIP 'alexandra' at 184.72.221.84:45357
Reliably Transmitting (NAT) to 184.72.221.84:45357:
OPTIONS sip:[email protected]:45357;rinstance=0171DB01;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK54e9653e;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as7856acf6
To: <sip:[email protected]:45357;rinstance=0171DB01;transport=tcp>
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert2
Date: Fri, 10 May 2013 09:39:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
       > Saved useragent "Acrobits SIPIS (http://www.acrobits.cz)" for peer alexandra

<--- Transmitting (NAT) to 184.72.221.84:45357 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.202.218.207:45357;branch=z9hG4bKHGgXr4J3tW3bG8xd;received=184.72.221.84;rport=45357
From: "alexandra" <sip:[email protected]>;tag=E22ADF38D9CB76615C44E37288997503
To: "alexandra" <sip:[email protected]>;tag=as5f241421
Call-ID: F223524B282ECDEC3199E2ECB40B13F1F1270F7D
CSeq: 297 REGISTER
Server: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 600
Contact: <sip:[email protected]:45357;rinstance=0171DB01;transport=tcp>;expires=600
Date: Fri, 10 May 2013 09:39:03 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'F223524B282ECDEC3199E2ECB40B13F1F1270F7D' in 32000 ms (Method: REGISTER)

<--- SIP read from TCP:184.72.221.84:45357 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK54e9653e;rport=5060;received=212.40.173.65
Contact: <sip:[email protected]:45357;transport=tcp>
From: "asterisk" <sip:[email protected]>;tag=as7856acf6
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
To: <sip:[email protected]:45357;rinstance=0171DB01;transport=tcp>
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE
Supported: replaces, path
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
[May 10 11:39:03] NOTICE[19760]: chan_sip.c:21107 handle_response_peerpoke: Peer 'alexandra' is now Reachable. (152ms / 2000ms)
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 92.50.92.73:5060:
OPTIONS sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4 SIP/2.0
Via: SIP/2.0/UDP 212.40.173.65:5060;branch=z9hG4bK2a9bd390;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as1097b039
To: <sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert2
Date: Fri, 10 May 2013 09:39:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:92.50.92.73:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.40.173.65:5060;branch=z9hG4bK2a9bd390;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as1097b039
To: <sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4>;tag=5C2770A4D4AB922D
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Contact: <sip:[email protected];uniq=39FF8947CA4463C0F5D580987CAA4>
User-Agent: AVM FRITZ!Box 6360 Cable (um) 85.05.28 (Oct 18 2012)
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: 204

v=0
o=user 5079590 5079590 IN IP4 92.50.92.73
s=call
c=IN IP4 92.50.92.73
t=0 0
m=audio 7078 RTP/AVP 8 0 101
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:7079
a=ptime:30
<------------->
--- (15 headers 11 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:92.50.92.73:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.50.92.73:5060;rport;branch=z9hG4bK08A3BEBD14F16390
From: <sip:[email protected]>;tag=FE5449670B25B504
To: <sip:[email protected]>;tag=as346e1e73
Call-ID: [email protected]
CSeq: 3563 BYE
Authorization: Digest username="GUS_FRITZ", realm="asterisk", nonce="77c37d24", uri="sip:[email protected]:5060", response="b7381c3330b85f0b25815454d382915c", algorithm=MD5
X-RTP-Stat: CS=333;PS=465;ES=515;OS=111600;SP=0/0;SO=0;QS=-;PR=2;ER=773;OR=160;CR=0;SR=0;QR=-;PL=0,0;BL=0;LS=0;RB=0/0;SB=-/-;EN=PCMA;DE=PCMA;JI=5,0;DL=0,0,0;IP=92.50.92.73:7098,92.50.92.74:64005
X-RTP-Stat-Add: DQ=1;DSS=0;DS=0;PLCS=111584;JS=0
Reason: Q.850; cause=16
Max-Forwards: 70
User-Agent: AVM FRITZ!Box 6360 Cable (um) 85.05.28 (Oct 18 2012)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 92.50.92.73:5060 (NAT)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 92.50.92.73:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.50.92.73:5060;branch=z9hG4bK08A3BEBD14F16390;received=92.50.92.73;rport=5060
From: <sip:[email protected]>;tag=FE5449670B25B504
To: <sip:[email protected]>;tag=as346e1e73
Call-ID: [email protected]
CSeq: 3563 BYE
Server: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:[email protected]:55278;transport=tcp> for address/port to send to
set_destination: set destination to 10.84.66.166:55278
Audio is at 10054
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 184.73.248.111:55278:
INVITE sip:[email protected]:55278;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK75001de3;rport
Max-Forwards: 70
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 270

v=0
o=root 2025264092 2025264094 IN IP4 212.40.173.65
s=Asterisk PBX 1.8.11-cert2
c=IN IP4 212.40.173.65
t=0 0
m=audio 10054 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Scheduling destruction of SIP dialog '[email protected]:5060' in 13696 ms (Method: INVITE)
  == Spawn extension (sipgate, 200, 3) exited non-zero on 'SIP/GUS_FRITZ-000009c4'

<--- SIP read from TCP:184.73.248.111:55278 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK75001de3;rport=5060;received=212.40.173.65
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
Call-ID: [email protected]:5060
CSeq: 104 INVITE
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Reliably Transmitting (NAT) to 217.10.68.150:5060:
OPTIONS sip:sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP 212.40.173.65:5060;branch=z9hG4bK56f55095;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as59db45ca
To: <sip:sipconnect.sipgate.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert2
Date: Fri, 10 May 2013 09:39:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.40.173.65:5060;branch=z9hG4bK56f55095;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as59db45ca
To: <sip:sipconnect.sipgate.de>;tag=fec8d079c35590678f285eba3d3e56d0.383b
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding: 
Accept-Language: en
Supported: 
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 212.117.203.34:5060:
OPTIONS sip:pro1.voipgateway.org SIP/2.0
Via: SIP/2.0/UDP 212.40.173.65:5060;branch=z9hG4bK27c1bc05;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as22a970d0
To: <sip:pro1.voipgateway.org>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert2
Date: Fri, 10 May 2013 09:39:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.117.203.34:5060 --->
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP 212.40.173.65:5060;branch=z9hG4bK27c1bc05;rport=5060
To: <sip:pro1.voipgateway.org>;tag=c176a244
From: "asterisk" <sip:[email protected]>;tag=as22a970d0
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from TCP:184.73.248.111:55278 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK75001de3;rport=5060;received=212.40.173.65
Contact: <sip:[email protected]:55278;transport=tcp>
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
Call-ID: [email protected]:5060
CSeq: 104 INVITE
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]:55278;transport=tcp> for address/port to send to
set_destination: set destination to 10.84.66.166:55278
Transmitting (NAT) to 184.73.248.111:55278:
ACK sip:[email protected]:55278;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 212.40.173.65:5060;branch=z9hG4bK75001de3;rport
Max-Forwards: 70
From: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
To: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Contact: <sip:[email protected]:5060;transport=TCP>
Call-ID: [email protected]:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.11-cert2
Content-Length: 0


---

<--- SIP read from TCP:184.73.248.111:55278 --->
BYE sip:[email protected]:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 10.84.66.166:55278;branch=z9hG4bKzKJIuctJrCsVSfQp;rport
Contact: <sip:[email protected]:55278;transport=tcp>
Max-Forwards: 70
From: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits SIPIS (http://www.acrobits.cz)
To: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
Call-ID: [email protected]:5060
CSeq: 1 BYE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 184.73.248.111:55278 (NAT)
Scheduling destruction of SIP dialog '[email protected]:5060' in 13696 ms (Method: BYE)

<--- Transmitting (NAT) to 184.73.248.111:55278 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.84.66.166:55278;branch=z9hG4bKzKJIuctJrCsVSfQp;received=184.73.248.111;rport=55278
From: <sip:[email protected]:55278;rinstance=149C5E2E;transport=tcp>;tag=603868585AE3AEEDC3F147788C5FEE25
To: "GUS_FRITZ" <sip:[email protected]>;tag=as7d4645c6
Call-ID: [email protected]:5060
CSeq: 1 BYE
Server: Asterisk PBX 1.8.11-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
gref03*CLI> sip set debug off

MaverrickTM · 10 Mai 2013

Ja, es scheint so als würde NAT verwendet werden.

rentier-s · 10 Mai 2013

Das scheint nicht nur so, die ganzen Audio-Pfade stimmen nicht. Und das nicht nur von einem Client.

Meine Tipps aus #14 hast Du wohl noch nicht umgesetzt, oder?

SIP-Einstellung für Port wird ignoriert (port 5060 eingestellt, aber Port 36112 wird

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