[Problem] Asterisk 14 - Verbindung bricht ab bei Rufannahme

SkyLin · 1 Dez 2016

Hallo,

ich setze gerade einen Asterisk 14 auf einem RaspberryPi3 mit Debian Jessie auf. Als Provider dient mir die Telekom mit ihrem VoIP. Zusätzlich habe ich zum Testen auf meinem Desktop Zoiper (6001) installiert und am Asterisk registriert. Mein Asterisk kann sich bei der Telekom registrieren. Ich kann Rufnummern nach aussen anrufen und mein Zoiper-Client klingelt auch. Nur jedes Mal wenn die andere Seite oder ich abhebe, wird die Verbindung beendet.
Auf meinem OpenWRT-Router ist nf-nat-sip und nf_conntrack_sip geladen, sowie für RTP die Portrange 30000-310000 von 217.0.23.4 (tel.t-online.de) nach 192.168.0.200 (Asterisk) geöffnet. Gleiches ist in der rtp.conf konfiguriert.
Hier meine pjsip.conf:

Code:

[global]
type=global
user_agent=Asterisk
endpoint_identifier_order=ip,username
default_from_user=rufnummer_mit_vorwahl

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.0.0/24

[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0
local_net=192.168.0.0/24

[telekom_rufnummer_mit_vorwahl]
type=registration
transport=transport-udp
outbound_auth=telekom_rufnummer_mit_vorwahl_auth
server_uri=sip:tel.t-online.de
client_uri=sip:[email protected]
contact_user=rufnummer_mit_vorwahl
retry_interval=60
forbidden_retry_interval=300
expiration=480
auth_rejection_permanent=false

[telekom_rufnummer_mit_vorwahl_auth]
type=auth
auth_type=userpass
password=secret
username=rufnummer_mit_vorwahl
realm=tel.t-online.de

[telekom_out]
type=endpoint
transport=transport-udp
context=unspecified
disallow=all
allow=g722
allow=alaw
outbound_auth=telekom_rufnummer_mit_vorwahl_auth
aors=telekom_out
callerid=rufnummer_mit_vorwahl
from_user=rufnummer_mit_vorwahl
from_domain=tel.t-online.de

[telekom_out]
type=aor
contact=sip:[email protected]

[telekom_in]
type=endpoint
transport=transport-udp
context=telekom_in
disallow=all
allow=g722
allow=alaw
outbound_auth=telekom_rufnummer_mit_vorwahl_auth

[telekom_in]
type=identify
endpoint=telekom_in
match=217.0.0.0/13

[6001]
type=endpoint
transport=transport-udp
context=internalsip
disallow=all
allow=g722
allow=alaw
auth=auth6001
aors=6001
mailboxes=wie in voicemail.conf definiert

[auth6001]
type=auth
auth_type=userpass
password=password
username=6001
realm=example.com

[6001]
type=aor
max_contacts=1
remove_existing=true

[6001]
type=identify
endpoint=6001
match=192.168.0.2

[acl]
type=acl
deny=0.0.0.0/0.0.0.0
; Telekom
permit=217.0.0.0/13
; eigenes LAN
permit=192.168.0.0/16

und extension.conf:

Code:

[general]
static=yes
writeprotect=yes
autofallthrough=yes
extenpatternmatchnew=no
clearglobalvars=no
userscontext=unspecified

[unspecified]
; wer hier landet ist entweder schlecht konfiguriert oder hat keine "Rechte"

exten => _X.,1,Answer()
exten => _X.,2,Verbose(D E F A U L T ==> ${CALLERID(num)} kam um ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} in UNSPECIFIED an als er versuchte die Nummer ${EXTEN} anzurufen.)
exten => _X.,3,Hangup()

[internalsip]
; in den Kontext gelangt man, wenn man einen Call
; von den internen Telefonen startet

; direkt einzelne User anwaehlen
exten => contact_name1,1,Dial(PJSIP/contact_name1)
exten => contact_name2,1,Dial(PJSIP/contact_name2)

;Mailboxabfrage von intern ohne PIN
; exten => mailboxname,1,VoiceMailMain(mailboxname@mailboxcontext,s)

;national, mit +49 gewaehlt
exten => _+49X.,1,Dial(PJSIP/telekom_out/sip:0${EXTEN:3}@tel.t-online.de,60)
exten => _+49X.,n,Hangup()

;international lassen wir nicht zu
exten => _+X.,1,Hangup() 
exten => _00X.,1,Hangup() 

;national, mit 0 vorneweg
exten => _0Z.,1,Dial(PJSIP/telekom_out/sip:${EXTEN}@tel.t-online.de,60)
exten => _0Z.,n,Hangup() 

; Ortsnetz
exten => _Z.,1,Dial(PJSIP/telekom_out/sip:Ortsnetzkennzahl-mit-0${EXTEN}@tel.t-online.de,60)
exten => _Z.,n,Hangup() 

; Notrufe gehen immer
exten => 110,1,Dial(PJSIP/telekom_out/sip:[email protected],60)
exten => 110,n,Hangup() 
exten => 112,1,Dial(PJSIP/telekom_out/sip:[email protected],60)
exten => 112,n,Hangup()

[telekom_in] 
; Anrufe von extern via Telekom
; 30 Sekunden klingen
exten => rufnummer_mit_vorwahl,1,Dial(PJSIP/6001,30) 
; danach auf die Mailbox umleiten
; exten => rufnummer_mit_vorwahl,n,VoiceMail(mailboxname@mailboxcontext)
exten => rufnummer_mit_vorwahl,n,Hangup()

Kann mir jemand weiterhelfen, warum die Verbindungen immer sofort abbrechen? Hier ist noch ein Auszug von der Asterisk-CLI:

Code:

[Dec  1 17:36:52] WARNING[1931]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
    -- Executing [rufnummer_mit_vorwahl@telekom_in:1] Dial("PJSIP/telekom_in-00000008", "PJSIP/6001,30") in new stack
    -- Called PJSIP/6001
    -- PJSIP/6001-00000009 is ringing
  == Spawn extension (telekom_in, rufnummer_mit_vorwahl, 1) exited non-zero on 'PJSIP/telekom_in-00000008'
[Dec  1 17:42:16] WARNING[1941]: res_pjsip_pubsub.c:3085 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Dec  1 17:42:16] WARNING[1942]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
    -- Executing [rufnummer_mit_vorwahl@telekom_in:1] Dial("PJSIP/telekom_in-0000000a", "PJSIP/6001,30") in new stack
    -- Called PJSIP/6001
    -- PJSIP/6001-0000000b is ringing
  == Spawn extension (telekom_in, rufnummer_mit_vorwahl, 1) exited non-zero on 'PJSIP/telekom_in-0000000a'

Gruß
SkyLin

rentier-s · 2 Dez 2016

Das ist oftmals ein Problem in der RTP Aushandlung. Mach mal ein SIP Debug und schau, was nach dem 180 Ringing passiert.

Die Config sieht eigentlich brauchbar aus, außer dass ich nicht direkt die SIP URI anwählen würde, sondern über den Endpoint, also Dial(PJSIP/${EXTEN}@telekom_out)

SkyLin · 2 Dez 2016

rentier-s schrieb:
Das ist oftmals ein Problem in der RTP Aushandlung. Mach mal ein SIP Debug und schau, was nach dem 180 Ringing passiert.

Ich habe einmal einen Anruf vom Zoiper-Client nach draussen debugged:

Code:

<--- Transmitting SIP response (534 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-3e0405535ddc9011-1---d8754z-
Call-ID: OWRhNzIxNDVlYjJjOWIwYmE0NTUwMGI2N2RmOTViNDQ.
From: <sip:[email protected]>;tag=cf9f9832
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-3e0405535ddc9011-1---d8754z-
CSeq: 1 PUBLISH
WWW-Authenticate: Digest  realm="example.com",nonce="1480734638/b360a872b43b37da7036ad65f4c04621",opaque="1aa4ffde48ed0308",algorithm=md5,qop="auth"
Server: Asterisk
Content-Length:  0

<--- Received SIP request (399 bytes) from UDP:192.168.0.2:35800 --->
ACK sip:+49angewä[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-e1293b0e99867faf-1---d8754z-
Max-Forwards: 70
To: <sip:+49angewä[email protected]>;tag=z9hG4bK-d8754z-e1293b0e99867faf-1---d8754z-
From: <sip:[email protected];transport=UDP>;tag=56383932
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 1 ACK
Content-Length: 0

<--- Transmitting SIP response (536 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-a246301f3103bcc1-1---d8754z-
Call-ID: YjAzOWE5NWFkZWQzOWE2NDc4ZGNiMGFlZDFkODNkYTQ.
From: <sip:[email protected]>;tag=d447902d
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-a246301f3103bcc1-1---d8754z-
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest  realm="example.com",nonce="1480734638/b360a872b43b37da7036ad65f4c04621",opaque="5dbe832977f2e422",algorithm=md5,qop="auth"
Server: Asterisk
Content-Length:  0

<--- Received SIP request (1228 bytes) from UDP:192.168.0.2:35800 --->
INVITE sip:+49angewä[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-8286e6786ff63f63-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:+49angewä[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=56383932
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6001",realm="example.com",nonce="1480734638/b360a872b43b37da7036ad65f4c04621",uri="sip:+49angewä[email protected];transport=UDP",response="8990f0fc055f50bd90bd91c6b8f535b2",cnonce="c8338c049700e654f3a0ab5063908849",nc=00000001,qop=auth,algorithm=md5,opaque="67da0d205f368982"
Allow-Events: presence, kpml
Content-Length: 237

v=0
o=Z 0 0 IN IP4 192.168.0.2
s=Z
c=IN IP4 192.168.0.2
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Received SIP request (1269 bytes) from UDP:192.168.0.2:35800 --->
PUBLISH sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-a64b155dc32112f8-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=cf9f9832
Call-ID: OWRhNzIxNDVlYjJjOWIwYmE0NTUwMGI2N2RmOTViNDQ.
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6001",realm="example.com",nonce="1480734638/b360a872b43b37da7036ad65f4c04621",uri="sip:[email protected];transport=UDP",response="fe8b34f85cc671bf28f46570739ec071",cnonce="de6bf47caade81f98c3ebfbd8a1828c1",nc=00000001,qop=auth,algorithm=md5,opaque="1aa4ffde48ed0308"
Event: presence
Allow-Events: presence, kpml
Content-Length: 267

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
          entity="sip:[email protected];transport=UDP">
  <tuple id="6001" >
     <status><basic>open</basic></status>
     <note>On the phone</note>
  </tuple>
</presence>

<--- Received SIP request (1011 bytes) from UDP:192.168.0.2:35800 --->
SUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-7851ddbe8e32b40c-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=d447902d
Call-ID: YjAzOWE5NWFkZWQzOWE2NDc4ZGNiMGFlZDFkODNkYTQ.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6001",realm="example.com",nonce="1480734638/b360a872b43b37da7036ad65f4c04621",uri="sip:[email protected];transport=UDP",response="cf991ac98b8d4055a4c683c97b6ac962",cnonce="bc71655dbf05a83c8331008960a56012",nc=00000001,qop=auth,algorithm=md5,opaque="5dbe832977f2e422"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

[Dec  3 04:10:38] WARNING[1946]: res_pjsip_pubsub.c:3085 pubsub_on_rx_publish_request: No registered publish handler for event presence
<--- Transmitting SIP response (378 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-a64b155dc32112f8-1---d8754z-
Call-ID: OWRhNzIxNDVlYjJjOWIwYmE0NTUwMGI2N2RmOTViNDQ.
From: <sip:[email protected]>;tag=cf9f9832
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-a64b155dc32112f8-1---d8754z-
CSeq: 2 PUBLISH
Server: Asterisk
Content-Length:  0

[Dec  3 04:10:38] WARNING[1946]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
<--- Transmitting SIP response (380 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-7851ddbe8e32b40c-1---d8754z-
Call-ID: YjAzOWE5NWFkZWQzOWE2NDc4ZGNiMGFlZDFkODNkYTQ.
From: <sip:[email protected]>;tag=d447902d
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-7851ddbe8e32b40c-1---d8754z-
CSeq: 2 SUBSCRIBE
Server: Asterisk
Content-Length:  0

<--- Transmitting SIP response (335 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-8286e6786ff63f63-1---d8754z-
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
From: <sip:[email protected]>;tag=56383932
To: <sip:+49angewä[email protected]>
CSeq: 2 INVITE
Server: Asterisk
Content-Length:  0

    -- Executing [+49angewählteNummer@internalsip:1] Dial("PJSIP/6001-00000006", "PJSIP/telekom_out/sip:0angewä[email protected],60") in new stack
    -- Called PJSIP/telekom_out/sip:0angewä[email protected]
<--- Transmitting SIP request (930 bytes) to UDP:217.0.20.236:5060 --->
INVITE sip:0angewä[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj7a350b37-6fd6-4ca2-94d7-6333058dd759
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
To: <sip:0angewä[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26392 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   261

v=0
o=- 749201700 749201700 IN IP4 192.168.0.200
s=Asterisk
c=IN IP4 192.168.0.200
t=0 0
m=audio 30028 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (561 bytes) from UDP:217.0.20.236:5060 --->
SIP/2.0 407 Proxy Authentication Required 02035034C
Via: SIP/2.0/UDP 192.168.0.200:5060;received=192.168.0.200;rport=5060;branch=z9hG4bKPj7a350b37-6fd6-4ca2-94d7-6333058dd759
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_314a88fa074fbb0e60c5cc963fe30927
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26392 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce="A3D1D0BEDC9441580000000004778F1D",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true

<--- Transmitting SIP request (434 bytes) to UDP:217.0.20.236:5060 --->
ACK sip:0angewä[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj7a350b37-6fd6-4ca2-94d7-6333058dd759
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_314a88fa074fbb0e60c5cc963fe30927
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26392 ACK
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

<--- Transmitting SIP request (1218 bytes) to UDP:217.0.20.236:5060 --->
INVITE sip:0angewä[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj8e42b50d-de7b-44f7-9f36-22f5174fa83c
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
To: <sip:0angewä[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26393 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Proxy-Authorization: Digest username="abgehendeNummer", realm="tel.t-online.de", nonce="A3D1D0BEDC9441580000000004778F1D", uri="sip:0angewä[email protected]", response="a72941f1f918b691eab7a4ddba41a9f5", algorithm=MD5, cnonce="a9291bf9-6fbf-48bf-bd9c-0681a9282ce2", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   261

v=0
o=- 749201700 749201700 IN IP4 192.168.0.200
s=Asterisk
c=IN IP4 192.168.0.200
t=0 0
m=audio 30028 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (353 bytes) from UDP:217.0.20.236:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.200:5060;received=192.168.0.200;rport=5060;branch=z9hG4bKPj8e42b50d-de7b-44f7-9f36-22f5174fa83c
To: <sip:0angewä[email protected]>
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26393 INVITE
Content-Length: 0

<--- Received SIP response (944 bytes) from UDP:217.0.20.236:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.200:5060;received=192.168.0.200;rport=5060;branch=z9hG4bKPj8e42b50d-de7b-44f7-9f36-22f5174fa83c
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26393 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.20.236;transport=udp;lr>
Require: 100rel
RSeq: 2
Supported: timer
Content-Type: application/sdp
Content-Length: 224
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

v=0
o=- 1852312853 2973781489 IN IP4 217.0.4.133
s=Basic Session
c=IN IP4 217.0.4.133
t=0 0
m=audio 49212 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<--- Transmitting SIP request (535 bytes) to UDP:217.0.20.236:5060 --->
PRACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj78040857-3773-4b67-b5f7-ae14bfcf55e5
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26394 PRACK
Route: <sip:217.0.20.236;transport=udp;lr>
RAck: 2 26393 INVITE
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

    -- PJSIP/telekom_out-00000007 is making progress passing it to PJSIP/6001-00000006
<--- Transmitting SIP response (784 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-8286e6786ff63f63-1---d8754z-
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
From: <sip:[email protected]>;tag=56383932
To: <sip:+49angewä[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
CSeq: 2 INVITE
Server: Asterisk
Contact: <sip:192.168.0.22:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Content-Type: application/sdp
Content-Length:   219

v=0
o=- 0 2 IN IP4 192.168.0.22
s=Asterisk
c=IN IP4 192.168.0.22
t=0 0
m=audio 30084 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

       > 0x7610ea80 -- Probation passed - setting RTP source address to 192.168.0.2:8000
       > 0x761192e0 -- Probation passed - setting RTP source address to 217.0.4.133:49212
<--- Received SIP response (528 bytes) from UDP:217.0.20.236:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;received=192.168.0.200;rport=5060;branch=z9hG4bKPj78040857-3773-4b67-b5f7-ae14bfcf55e5
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26394 PRACK
Content-Length: 0
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

<--- Received SIP response (935 bytes) from UDP:217.0.20.236:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.200:5060;received=192.168.0.200;rport=5060;branch=z9hG4bKPj8e42b50d-de7b-44f7-9f36-22f5174fa83c
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26393 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.20.236;transport=udp;lr>
Require: 100rel
RSeq: 3
Supported: timer
Content-Type: application/sdp
Content-Length: 224
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

v=0
o=- 1852312853 2973781489 IN IP4 217.0.4.133
s=Basic Session
c=IN IP4 217.0.4.133
t=0 0
m=audio 49212 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<--- Transmitting SIP request (535 bytes) to UDP:217.0.20.236:5060 --->
PRACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj14235a04-a2a2-4c20-81a7-f4f388dc7c77
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26395 PRACK
Route: <sip:217.0.20.236;transport=udp;lr>
RAck: 3 26393 INVITE
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

    -- PJSIP/telekom_out-00000007 is ringing
<--- Transmitting SIP response (775 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-8286e6786ff63f63-1---d8754z-
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
From: <sip:[email protected]>;tag=56383932
To: <sip:+49angewä[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
CSeq: 2 INVITE
Server: Asterisk
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Contact: <sip:192.168.0.22:5060>
Content-Type: application/sdp
Content-Length:   219

v=0
o=- 0 2 IN IP4 192.168.0.22
s=Asterisk
c=IN IP4 192.168.0.22
t=0 0
m=audio 30084 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

    -- PJSIP/6001-00000006 requested media update control 26, passing it to PJSIP/telekom_out-00000007
<--- Received SIP response (528 bytes) from UDP:217.0.20.236:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;received=192.168.0.200;rport=5060;branch=z9hG4bKPj14235a04-a2a2-4c20-81a7-f4f388dc7c77
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26395 PRACK
Content-Length: 0
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

<--- Received SIP response (1180 bytes) from UDP:217.0.20.236:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;received=192.168.0.200;rport=5060;branch=z9hG4bKPj8e42b50d-de7b-44f7-9f36-22f5174fa83c
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26393 INVITE
Contact: <sip:[email protected];transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Record-Route: <sip:217.0.20.236;transport=udp;lr>
Session-Expires: 1800;refresher=uas
Supported: timer
Content-Type: application/sdp
Content-Length: 224
Session-ID: a8ffc339dfba1b39413020fb7703ee2f
Authentication-Info: qop=auth,rspauth="fd21439be8d257413e18af67c9d9af6a",cnonce="a9291bf9-6fbf-48bf-bd9c-0681a9282ce2",nc=00000001
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

v=0
o=- 1852312853 2973781489 IN IP4 217.0.4.133
s=Basic Session
c=IN IP4 217.0.4.133
t=0 0
m=audio 49212 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<--- Transmitting SIP request (509 bytes) to UDP:217.0.20.236:5060 --->
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj15328ddf-e8ad-46e1-b21a-2bd1ba5ca269
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26393 ACK
Route: <sip:217.0.20.236;transport=udp;lr>
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

    -- PJSIP/telekom_out-00000007 answered PJSIP/6001-00000006
<--- Transmitting SIP response (818 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-8286e6786ff63f63-1---d8754z-
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
From: <sip:[email protected]>;tag=56383932
To: <sip:+49angewä[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
CSeq: 2 INVITE
Server: Asterisk
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Contact: <sip:192.168.0.22:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   219

v=0
o=- 0 2 IN IP4 192.168.0.22
s=Asterisk
c=IN IP4 192.168.0.22
t=0 0
m=audio 30084 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

    -- Channel PJSIP/telekom_out-00000007 joined 'simple_bridge' basic-bridge <c583b77d-617f-4ed1-9486-d8119a42922e>
    -- Channel PJSIP/6001-00000006 joined 'simple_bridge' basic-bridge <c583b77d-617f-4ed1-9486-d8119a42922e>
       > Bridge c583b77d-617f-4ed1-9486-d8119a42922e: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'PJSIP/6001-00000006' and 'PJSIP/telekom_out-00000007' - media will flow directly between them
       > Remotely bridged 'PJSIP/6001-00000006' and 'PJSIP/telekom_out-00000007' - media will flow directly between them
<--- Received SIP request (752 bytes) from UDP:192.168.0.2:35800 --->
ACK sip:192.168.0.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-fae5dec376a37ff2-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:+49angewä[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
From: <sip:[email protected]>;tag=56383932
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 2 ACK
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6001",realm="example.com",nonce="1480734638/b360a872b43b37da7036ad65f4c04621",uri="sip:+49angewä[email protected];transport=UDP",response="8990f0fc055f50bd90bd91c6b8f535b2",cnonce="c8338c049700e654f3a0ab5063908849",nc=00000001,qop=auth,algorithm=md5,opaque="67da0d205f368982"
Content-Length: 0

<--- Transmitting SIP request (891 bytes) to UDP:192.168.0.2:35800 --->
INVITE sip:[email protected]:35800;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;rport;branch=z9hG4bKPjb2003b0d-35de-4d1f-b3c8-04ccf1c17f75
From: <sip:[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
To: <sip:[email protected]>;tag=56383932
Contact: <sip:192.168.0.22:5060>
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 5012 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   219

v=0
o=- 0 3 IN IP4 192.168.0.200
s=Asterisk
c=IN IP4 217.0.4.133
t=0 0
m=audio 49212 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (1038 bytes) to UDP:217.0.20.236:5060 --->
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj2b9d1b12-71ef-4f8c-9813-a73e87484fa9
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
Contact: <sip:[email protected]:5060>
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26396 INVITE
Route: <sip:217.0.20.236;transport=udp;lr>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   234

v=0
o=- 749201700 749201701 IN IP4 192.168.0.200
s=Asterisk
c=IN IP4 192.168.0.2
t=0 0
m=audio 8000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (942 bytes) from UDP:192.168.0.2:35800 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.22:5060;rport=5060;branch=z9hG4bKPjb2003b0d-35de-4d1f-b3c8-04ccf1c17f75
Require: timer
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected]>;tag=56383932
From: <sip:[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 5012 INVITE
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 237

v=0
o=Z 0 1 IN IP4 192.168.0.2
s=Z
c=IN IP4 192.168.0.2
t=0 0
m=audio 8000 RTP/AVP 8 3 110 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP request (405 bytes) to UDP:192.168.0.2:35800 --->
ACK sip:[email protected]:35800;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;rport;branch=z9hG4bKPj658bfc97-cdab-429e-9de5-126d76113cb3
From: <sip:[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
To: <sip:[email protected]>;tag=56383932
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 5012 ACK
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

<--- Received SIP request (965 bytes) from UDP:192.168.0.2:35800 --->
PUBLISH sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-1b99c6e5b8e59563-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=a57c2728
Call-ID: MDMyMDFjNzBkOGU0M2U4Y2U3YTY3NmI5ZWNiOWRhOTY.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 267

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
          entity="sip:[email protected];transport=UDP">
  <tuple id="6001" >
     <status><basic>open</basic></status>
     <note>On the phone</note>
  </tuple>
</presence>

<--- Received SIP request (707 bytes) from UDP:192.168.0.2:35800 --->
SUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-51ef788c55dc5ab0-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=b40c2f10
Call-ID: MDM1MTMwY2M1YWYyNTJiYTgzZjVlY2FkYzlmNmVkY2E.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<--- Transmitting SIP response (534 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-1b99c6e5b8e59563-1---d8754z-
Call-ID: MDMyMDFjNzBkOGU0M2U4Y2U3YTY3NmI5ZWNiOWRhOTY.
From: <sip:[email protected]>;tag=a57c2728
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-1b99c6e5b8e59563-1---d8754z-
CSeq: 1 PUBLISH
WWW-Authenticate: Digest  realm="example.com",nonce="1480734649/2aa09ac71bc3345b503377444e267a32",opaque="0de358d64500309d",algorithm=md5,qop="auth"
Server: Asterisk
Content-Length:  0

<--- Transmitting SIP response (536 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-51ef788c55dc5ab0-1---d8754z-
Call-ID: MDM1MTMwY2M1YWYyNTJiYTgzZjVlY2FkYzlmNmVkY2E.
From: <sip:[email protected]>;tag=b40c2f10
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-51ef788c55dc5ab0-1---d8754z-
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest  realm="example.com",nonce="1480734649/2aa09ac71bc3345b503377444e267a32",opaque="09d2d5dc6ee08f4c",algorithm=md5,qop="auth"
Server: Asterisk
Content-Length:  0

<--- Received SIP request (1269 bytes) from UDP:192.168.0.2:35800 --->
PUBLISH sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-2126f49a9bf2b071-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=a57c2728
Call-ID: MDMyMDFjNzBkOGU0M2U4Y2U3YTY3NmI5ZWNiOWRhOTY.
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6001",realm="example.com",nonce="1480734649/2aa09ac71bc3345b503377444e267a32",uri="sip:[email protected];transport=UDP",response="ba2387d929f91d8873257037951e556c",cnonce="d9964361744c0439abfb67bd2c8cef4d",nc=00000001,qop=auth,algorithm=md5,opaque="0de358d64500309d"
Event: presence
Allow-Events: presence, kpml
Content-Length: 267

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
          entity="sip:[email protected];transport=UDP">
  <tuple id="6001" >
     <status><basic>open</basic></status>
     <note>On the phone</note>
  </tuple>
</presence>

<--- Received SIP request (1011 bytes) from UDP:192.168.0.2:35800 --->
SUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-9e59bbee3b2cbe83-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=b40c2f10
Call-ID: MDM1MTMwY2M1YWYyNTJiYTgzZjVlY2FkYzlmNmVkY2E.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6001",realm="example.com",nonce="1480734649/2aa09ac71bc3345b503377444e267a32",uri="sip:[email protected];transport=UDP",response="bed9ff1176cb3e68b28948ef11d1a130",cnonce="a1aa08179843dfd7e5e4ed95edeb462e",nc=00000001,qop=auth,algorithm=md5,opaque="09d2d5dc6ee08f4c"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

[Dec  3 04:10:49] WARNING[1945]: res_pjsip_pubsub.c:3085 pubsub_on_rx_publish_request: No registered publish handler for event presence
<--- Transmitting SIP response (378 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-2126f49a9bf2b071-1---d8754z-
Call-ID: MDMyMDFjNzBkOGU0M2U4Y2U3YTY3NmI5ZWNiOWRhOTY.
From: <sip:[email protected]>;tag=a57c2728
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-2126f49a9bf2b071-1---d8754z-
CSeq: 2 PUBLISH
Server: Asterisk
Content-Length:  0

[Dec  3 04:10:49] WARNING[1946]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
<--- Transmitting SIP response (380 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-9e59bbee3b2cbe83-1---d8754z-
Call-ID: MDM1MTMwY2M1YWYyNTJiYTgzZjVlY2FkYzlmNmVkY2E.
From: <sip:[email protected]>;tag=b40c2f10
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-9e59bbee3b2cbe83-1---d8754z-
CSeq: 2 SUBSCRIBE
Server: Asterisk
Content-Length:  0

<--- Received SIP response (428 bytes) from UDP:217.0.20.236:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.200:5060;received=192.168.0.200;rport=5060;branch=z9hG4bKPj2b9d1b12-71ef-4f8c-9813-a73e87484fa9
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26396 INVITE
Content-Length: 0

<--- Transmitting SIP request (509 bytes) to UDP:217.0.20.236:5060 --->
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj2b9d1b12-71ef-4f8c-9813-a73e87484fa9
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26396 ACK
Route: <sip:217.0.20.236;transport=udp;lr>
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

<--- Transmitting SIP request (533 bytes) to UDP:217.0.20.236:5060 --->
BYE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPj17c11310-63ab-4192-b203-bd7264104bbf
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26397 BYE
Route: <sip:217.0.20.236;transport=udp;lr>
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

<--- Received SIP response (526 bytes) from UDP:217.0.20.236:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;received=192.168.0.200;rport=5060;branch=z9hG4bKPj17c11310-63ab-4192-b203-bd7264104bbf
To: <sip:0angewä[email protected]>;tag=h7g4Esbg_p65551t1480692945m197137c420781843s1_2969902511-1017950774
From: <sip:[email protected]>;tag=524474d3-ab8c-41ff-a187-5d7e03a99447
Call-ID: d7a8285e-5e3e-4774-a605-abd4cfeef0b8
CSeq: 26397 BYE
Content-Length: 0
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

    -- Channel PJSIP/telekom_out-00000007 left 'native_rtp' basic-bridge <c583b77d-617f-4ed1-9486-d8119a42922e>
    -- Channel PJSIP/6001-00000006 left 'native_rtp' basic-bridge <c583b77d-617f-4ed1-9486-d8119a42922e>
<--- Transmitting SIP request (905 bytes) to UDP:192.168.0.2:35800 --->
INVITE sip:[email protected]:35800;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;rport;branch=z9hG4bKPj25a73d82-8e50-4b60-8433-4298f11ebb05
From: <sip:[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
To: <sip:[email protected]>;tag=56383932
Contact: <sip:192.168.0.22:5060>
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 5013 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   219

v=0
o=- 0 4 IN IP4 192.168.0.22
s=Asterisk
c=IN IP4 192.168.0.22
t=0 0
m=audio 30084 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

  == Spawn extension (internalsip, +49angewählteNummer, 1) exited non-zero on 'PJSIP/6001-00000006'
<--- Received SIP response (942 bytes) from UDP:192.168.0.2:35800 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.22:5060;rport=5060;branch=z9hG4bKPj25a73d82-8e50-4b60-8433-4298f11ebb05
Require: timer
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected]>;tag=56383932
From: <sip:[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 5013 INVITE
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 237

v=0
o=Z 0 2 IN IP4 192.168.0.2
s=Z
c=IN IP4 192.168.0.2
t=0 0
m=audio 8000 RTP/AVP 8 3 110 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP request (405 bytes) to UDP:192.168.0.2:35800 --->
ACK sip:[email protected]:35800;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;rport;branch=z9hG4bKPj7ef1616f-f335-4f5e-b714-b36802f57abd
From: <sip:[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
To: <sip:[email protected]>;tag=56383932
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 5013 ACK
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

<--- Transmitting SIP request (404 bytes) to UDP:192.168.0.2:35800 --->
BYE sip:[email protected]:35800;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:5060;rport;branch=z9hG4bKPj6b865e42-56c0-42c0-9ca4-efa7330d51a6
From: <sip:[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
To: <sip:[email protected]>;tag=56383932
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 5014 BYE
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

<--- Received SIP response (415 bytes) from UDP:192.168.0.2:35800 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.22:5060;rport=5060;branch=z9hG4bKPj6b865e42-56c0-42c0-9ca4-efa7330d51a6
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected]>;tag=56383932
From: <sip:[email protected]>;tag=113814b1-f089-4ca7-806c-e58c939686d2
Call-ID: MGQyOTYzZTIyMzA4MWMyYzFmNzAzMmI1MGY2MTQ4ZDI.
CSeq: 5014 BYE
User-Agent: Z 3.3.25608 r25552
Content-Length: 0

<--- Received SIP request (959 bytes) from UDP:192.168.0.2:35800 --->
PUBLISH sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-341d2a8b955df2ea-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=ffc6977b
Call-ID: Nzc1M2YzNDRiMDJmYzc3YTNjZGI1YTZhMmFiMzkxZWM.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 261

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
          entity="sip:[email protected];transport=UDP">
  <tuple id="6001" >
     <status><basic>open</basic></status>
     <note>Online</note>
  </tuple>
</presence>

<--- Received SIP request (707 bytes) from UDP:192.168.0.2:35800 --->
SUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-0861d5e45642a422-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=accc7612
Call-ID: OTNiMmFjOTZlNDNlNTRmZTE1NzliMjdhZTgxNTY4MWU.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<--- Transmitting SIP response (534 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-341d2a8b955df2ea-1---d8754z-
Call-ID: Nzc1M2YzNDRiMDJmYzc3YTNjZGI1YTZhMmFiMzkxZWM.
From: <sip:[email protected]>;tag=ffc6977b
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-341d2a8b955df2ea-1---d8754z-
CSeq: 1 PUBLISH
WWW-Authenticate: Digest  realm="example.com",nonce="1480734649/2aa09ac71bc3345b503377444e267a32",opaque="129e3c6f0ed283ff",algorithm=md5,qop="auth"
Server: Asterisk
Content-Length:  0

<--- Transmitting SIP response (536 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-0861d5e45642a422-1---d8754z-
Call-ID: OTNiMmFjOTZlNDNlNTRmZTE1NzliMjdhZTgxNTY4MWU.
From: <sip:[email protected]>;tag=accc7612
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-0861d5e45642a422-1---d8754z-
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest  realm="example.com",nonce="1480734649/2aa09ac71bc3345b503377444e267a32",opaque="3b8df78b22f60cb9",algorithm=md5,qop="auth"
Server: Asterisk
Content-Length:  0

<--- Received SIP request (1263 bytes) from UDP:192.168.0.2:35800 --->
PUBLISH sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-01c5c64d3cef98a6-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=ffc6977b
Call-ID: Nzc1M2YzNDRiMDJmYzc3YTNjZGI1YTZhMmFiMzkxZWM.
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6001",realm="example.com",nonce="1480734649/2aa09ac71bc3345b503377444e267a32",uri="sip:[email protected];transport=UDP",response="aaefa2181913938d376bf8a869cdfc3a",cnonce="0a2de4adf778cc0ece2616674af428c4",nc=00000001,qop=auth,algorithm=md5,opaque="129e3c6f0ed283ff"
Event: presence
Allow-Events: presence, kpml
Content-Length: 261

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
          entity="sip:[email protected];transport=UDP">
  <tuple id="6001" >
     <status><basic>open</basic></status>
     <note>Online</note>
  </tuple>
</presence>

<--- Received SIP request (1011 bytes) from UDP:192.168.0.2:35800 --->
SUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:35800;branch=z9hG4bK-d8754z-22f8ea9f3dbaa4c5-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:35800;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=accc7612
Call-ID: OTNiMmFjOTZlNDNlNTRmZTE1NzliMjdhZTgxNTY4MWU.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username="6001",realm="example.com",nonce="1480734649/2aa09ac71bc3345b503377444e267a32",uri="sip:[email protected];transport=UDP",response="48a281d4fa13debdeeaa3cb27d7d5f0e",cnonce="d9358502150f09e779f06e0d6d3c9a4f",nc=00000001,qop=auth,algorithm=md5,opaque="3b8df78b22f60cb9"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

[Dec  3 04:10:49] WARNING[1945]: res_pjsip_pubsub.c:3085 pubsub_on_rx_publish_request: No registered publish handler for event presence
<--- Transmitting SIP response (378 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-01c5c64d3cef98a6-1---d8754z-
Call-ID: Nzc1M2YzNDRiMDJmYzc3YTNjZGI1YTZhMmFiMzkxZWM.
From: <sip:[email protected]>;tag=ffc6977b
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-01c5c64d3cef98a6-1---d8754z-
CSeq: 2 PUBLISH
Server: Asterisk
Content-Length:  0

[Dec  3 04:10:49] WARNING[1946]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
<--- Transmitting SIP response (380 bytes) to UDP:192.168.0.2:35800 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.0.2:35800;rport=35800;received=192.168.0.2;branch=z9hG4bK-d8754z-22f8ea9f3dbaa4c5-1---d8754z-
Call-ID: OTNiMmFjOTZlNDNlNTRmZTE1NzliMjdhZTgxNTY4MWU.
From: <sip:[email protected]>;tag=accc7612
To: <sip:[email protected]>;tag=z9hG4bK-d8754z-22f8ea9f3dbaa4c5-1---d8754z-
CSeq: 2 SUBSCRIBE
Server: Asterisk
Content-Length:  0

Ehrlich gesagt, was ich noch nicht genau, was hier hier unnormal ist - wo ich hinschauen soll. Bzw. woran erkenne ich, dass die RTP-Aushandlung korrekt war?
Was mir auffällt ist "from UDP:217.0.20.236:5060". In OpenWRT habe ich 217.0.23.4 erlaubt. Inzwischen habe ich auch 217.0.20.236 erlaubt, ohne Erfolg.

rentier-s · 6 Dez 2016

Was mir auffällt ist "from UDP:217.0.20.236:5060". In OpenWRT habe ich 217.0.23.4 erlaubt. Inzwischen habe ich auch 217.0.20.236 erlaubt, ohne Erfolg.

Du brauchst auf jeden Fall das komplette Netzwerk der Telekom, ist irgendwas um die 217.0.0.0/13 (?)

Was genau versuchst Du da? Rufst Du Dich selber über Deine externe Nummer an? Da passieren nämlich komische Invites, die irgendwie auf eine Loop hindeuten. Jedenfalls geht das ganze mit einem 403 auf die Bretter.

Außerdem solltest Du zumindest für die externen endpoints direct_media=no setzen. Den Rest müsste das ALG (ist nf-nat-sip ein ALG?) hinbekommen.

SkyLin · 6 Dez 2016

rentier-s schrieb:
Du brauchst auf jeden Fall das komplette Netzwerk der Telekom, ist irgendwas um die 217.0.0.0/13 (?)

Das habe ich am Wochende auf der Firewall (OpenWRT/LUCI) bereits so geändert. Allerdings ändert das nichts an den Verbindungsabbrüchen.

rentier-s schrieb:
Was genau versuchst Du da? Rufst Du Dich selber über Deine externe Nummer an? Da passieren nämlich komische Invites, die irgendwie auf eine Loop hindeuten. Jedenfalls geht das ganze mit einem 403 auf die Bretter.

Na ja, so klug bin ich jetzt auch wieder nicht. Zum Testen der Festnetznummern - abgehend wie eingehend - nutze ich mein Smartphone, das nicht mit dem Asterisk verbandelt ist.

rentier-s schrieb:
Außerdem solltest Du zumindest für die externen endpoints direct_media=no setzen. Den Rest müsste das ALG (ist nf-nat-sip ein ALG?) hinbekommen.

Ob nf-nat-sip ein Application-level gateway ist, weiß ich nicht. Ich kenne es als Kernelmodul für Netfilter, um die Schwachstellen bei SIP des normalen NAT auszubügeln.

Code:

tcpdump -i pppoe-wan -s 65000 -w Sip-Session -nXv udp and port 5060

sowohl für die WAN-Schnittstelle wie an der LAN-Schnittstelle zeigt mir.

Code:

1    0.000000    217.0.23.103    217.83.128.240    SIP/SDP    1295    Request: INVITE sip:[email protected]:5060 | 
2    0.009614    217.83.128.240    217.0.23.103    SIP    511    Status: 100 Trying | 
3    0.229252    217.83.128.240    217.0.23.103    SIP    700    Status: 180 Ringing | 
4    4.401306    217.83.128.240    217.0.43.129    DNS    81    Standard query 0x9fba  A apkrep.ff.avast.com
.
.
.
20    9.382577    217.83.128.240    217.0.23.103    SIP/SDP    1068    Status: 200 OK | 
21    9.389391    217.83.128.240    217.0.5.70    RTP    216    PT=ITU-T G.711 PCMA, SSRC=0x5EB8690F, Seq=31382, Time=2038008616
.
.
.
31    9.596936    217.0.23.103    217.83.128.240    SIP    606    Request: ACK sip:217.83.128.240:5060 | 
32    9.601453    217.83.128.240    217.0.23.103    SIP/SDP    1096    Request: INVITE sip:[email protected];transport=udp, in-dialog | 
33    9.603117    217.83.128.240    217.0.5.70    RTP    216    PT=ITU-T G.711 PCMA, SSRC=0xDEFBB91B, Seq=43922, Time=608682618
.
.
.
37    9.630842    217.0.5.70    217.83.128.240    RTCP    120    Receiver Report   Source description   Goodbye   
38    9.643103    217.83.128.240    217.0.5.70    RTP    216    PT=ITU-T G.711 PCMA, SSRC=0xDEFBB91B, Seq=43924, Time=608682938
.
.
.
42    9.679065    217.0.23.103    217.83.128.240    SIP    495    Status: 403 Forbidden | 
43    9.681018    217.83.128.240    217.0.23.103    SIP    580    Request: ACK sip:[email protected];transport=udp | 
44    9.683140    217.83.128.240    217.0.23.103    SIP    604    Request: BYE sip:[email protected];transport=udp | 
45    9.683825    217.83.128.240    217.0.5.70    RTP    216    PT=ITU-T G.711 PCMA, SSRC=0xDEFBB91B, Seq=43926, Time=608683258
.
.
.
52    9.749847    217.0.23.103    217.83.128.240    SIP    598    Status: 200 OK | 
53    9.768449    217.83.128.240    217.0.5.70    RTCP    108    Sender Report   Source description   Goodbye

So sieht der Mitschnitt aus. Ohne Dein " endpoints direct_media=no", das möchte ich noch verstehen.

- - - Aktualisiert - - -

Also das Hinzufügen von "direct_media=no" zu allen Endpoints in der pjsip.conf hat bewirkt, dass zumindest die Verbindung nicht abgerissen ist und ein Telefonieren möglich war. Wie sich das im Produktivbetrieb auswirkt, muss ich jetzt einmal testen.

Bis dahin jedenfalls vielen Dank.

SkyLin

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