Verzögerung bei ausgehenden Anrufen

Lomas · 28 Apr 2022

Seit kurzem haben wir bei Anrufen folgenden Effekt: Beim Anrufer kommt nahezu sofort "Anruf beendet", nach ca. 10s klingelt es dann beim Angerufenen. Wenn der Angerufene dann abhebt, hört er natürlich nichts. Unser Asterisk befindet sich hinter einem IPFire. Der Effekt tritt aber sowohl bei ausgehenden als auch bei internen Anrufen auf. Aber auch nicht immer, manchmal funktioniert es auch. Eingehende Anrufe funktionieren dagegen immer ohne Probleme. Angemerkt sei, dass weder beim Asterisk noch auf der Firewall irgendwelche Updates eingespielt oder Konfigurationsänderungen vorgenommen wurden. Im Asterisk-Log finden sich Einträge wie

Code:

chan_sip.c: Hanging up call fqz35qc98@<asterisk-ip> - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Was ist die Ursache dafür bzw. wie kann ich das beheben?

Vielen Dank im Voraus.
Lomas

koyaanisqatsi · 28 Apr 2022

Wenn du des Englischen mächtig bist, steht alles im Link zur Fehlermeldung...

Home - Asterisk Documentation

wiki.asterisk.org

Eine Fernanalyse ist unmöglich anhand der Informationen deines Posts.
...da bedarf es mehr Infos zu deinem NAT und zu deiner Asterisk Konfiguration.

Wird zum Beispiel ICE/STUN genutzt?
( sip show settings )
...oder kann dein Asterisk direkt aus dem Internet erreicht werden?
Was steht im SIP? Wenn so was passiert.
( *-Konsole: sip set debug on )

Erst dann kann es mit Tipps losgehen.
Alles andere ist nur Raterei und mündet positiverweise höchstens in: "Infos aus der Nase ziehen"

Lomas · 28 Apr 2022

Die Settings:

Code:

Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    No
  Allow unknown access:   No
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Path support :          No
  Realm. auth:            No
  Our auth realm          asterisk
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 13.14.1~dfsg-2+deb9u3
  SDP Session Name:       Asterisk PBX 13.14.1~dfsg-2+deb9u3
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             No
  Send RPID:              No
  Legacy userfield parse: No
  Send Diversion:         Yes
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          4294967295
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No

Network Settings:
---------------------------
  SIP address remapping:  Enabled using externaddr
  Externhost:             <none>
  Externaddr:             77.24.110.145:0
  Externrefresh:          10
  Localnet:               192.168.63.0/255.255.255.0

Global Signalling Settings:
---------------------------
  Codecs:                 (ulaw|alaw|gsm|h263)
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          Yes
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  1800 secs
  Sub. min duration       60 secs
  Sub. max duration:      3600 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Outbound reg. retry 403:No
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set>
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     No
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                unauthenticated
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            No
  DTMF:                   rfc2833
  Qualify:                2000
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        Yes
  Language:               de
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk

Die sip.conf:

Code:

[general]

language          = de                ; Default language setting for all users/peers
context           = unauthenticated
allowguest        = no               
udpbindaddr       = 0.0.0.0           ; alle Schnittstellen
tcpenable         = no                ; Enable server for incoming TCP connections (default is no)
transport         = udp               ; Set the default transports.  The order determines the primary default transport.
srvlookup         = no                ; Enable DNS SRV lookups on outbound calls
maxexpirey        = 3600
defaultexpirey    = 1800
externip          = <externip>
localnet          = 192.168.63.0/255.255.255.0
directmedia       = no
qualify           = yes
prematuremedia    = no
progressinband    = yes
nat               = rport,comedia

[Easybell]
type         = peer
context      = easybell-in
fromuser     = Easybell
defaultuser  = 00xxxxxxxxxxx
username     = 00xxxxxxxxxxx
remotesecret = yyyyyyyyyyy
host         = sip.easybell.de
fromdomain   = sip.easybell.de
insecure     = port,invite
caninvite    = no
canreinvite  = no
disallow     = all
allow        = ulaw
allow        = alaw

Der Asterisk ist nicht direkt aus dem Internet erreichbar. In der Firewall sind Port 5060 sowie die Ports 10000-20000 freigegeben. Den Wiki-Artikel kenne ich. Mich wundert, dass das Problem auch bei internen Anrufen auftritt. Hier sollten doch NAT und Firewall-Regeln keine Rolle spielen, oder? Wenn die Anrufe funktionieren, kommen auch im CLI sofort die Ausgaben. Wenn der Anruf nicht funktioniert, passiert auch im CLI nichts. Erst nach den 10s Verzögerung erscheinen dann die Ausgaben im CLI.

Lomas · 28 Apr 2022

Und hier noch ein Mitschnitt, wenn es nicht funktioniert:

Code:

<--- SIP read from UDP:<snomip>:5060 --->
INVITE sip:+43xxxxxxxxx@<asterisk-ip>;user=phone SIP/2.0
Via: SIP/2.0/UDP <snomip>:5060;rport;branch=z9hG4bKy1y1rwlt92ab65ko
Max-Forwards: 70
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41672 INVITE
Contact: <sip:11@<snomip>;line=59723>
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Content-Disposition: session
Min-SE: 90
Session-Expires: 3600
Supported: replaces,100rel,timer
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Type: application/sdp
Content-Length: 288

v=0
o=11 332833410 332833410 IN IP4 <snomip>
s=-
c=IN IP4 <snomip>
t=0 0
m=audio 50032 RTP/AVP 0 8 115 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:80
a=sendrecv
a=rtcp:50033
<------------->
--- (16 headers 15 lines) ---
Sending to <snomip>:5060 (no NAT)
Sending to <snomip>:5060 (no NAT)
Using INVITE request as basis request - .w5c8kw7kdhrd833f@<asterisk-ip>
Found peer '11' for '11' from <snomip>:5060

<--- Reliably Transmitting (no NAT) to <snomip>:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKy1y1rwlt92ab65ko;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as0ee3b7c7
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41672 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="408a5310"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '.w5c8kw7kdhrd833f@<asterisk-ip>' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:<snomip>:5060 --->
ACK sip:+43xxxxxxxxx@<asterisk-ip>;user=phone SIP/2.0
Via: SIP/2.0/UDP <snomip>:5060;rport;branch=z9hG4bKy1y1rwlt92ab65ko
Max-Forwards: 70
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as0ee3b7c7
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41672 ACK
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Length: 0

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

<--- SIP read from UDP:<snomip>:5060 --->
INVITE sip:+43xxxxxxxxx@<asterisk-ip>;user=phone SIP/2.0
Via: SIP/2.0/UDP <snomip>:5060;rport;branch=z9hG4bKivuu07lcg24vgak
Max-Forwards: 70
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Contact: <sip:11@<snomip>;line=59723>
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Authorization: Digest username="11", realm="asterisk", nonce="408a5310", uri="sip:+43xxxxxxxxx@<asterisk-ip>;user=phone", response="46d9a71564327c3ba5dcea3cb597a0f7", algorithm=MD5
Content-Disposition: session
Min-SE: 90
Session-Expires: 3600
Supported: replaces,100rel,timer
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Type: application/sdp
Content-Length: 288

v=0
o=11 332833410 332833410 IN IP4 <snomip>
s=-
c=IN IP4 <snomip>
t=0 0
m=audio 50032 RTP/AVP 0 8 115 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:80
a=sendrecv
a=rtcp:50033
<------------->
--- (17 headers 15 lines) ---
Sending to <snomip>:5060 (no NAT)
Using INVITE request as basis request - .w5c8kw7kdhrd833f@<asterisk-ip>
Found peer '11' for '11' from <snomip>:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 115
Found RTP audio format 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 115
Found audio description format G729 for ID 18
Capabilities: us - (ulaw|alaw|g726|g722), peer - audio=(ulaw|alaw|g729|g726)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
       > 0x742062d8 -- Strict RTP learning after remote address set to: <snomip>:50032
Peer audio RTP is at port <snomip>:50032
Looking for +43xxxxxxxxx in from-internal (domain <asterisk-ip>)
sip_route_dump: route/path hop: <sip:11@<snomip>;line=59723>

<--- Transmitting (no NAT) to <snomip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Length: 0


<------------>
    -- Executing [+43xxxxxxxxx@from-internal:1] NoOp("SIP/11-00000078", "") in new stack

<--- SIP read from UDP:<snomip>:5060 --->
INVITE sip:+43xxxxxxxxx@<asterisk-ip>;user=phone SIP/2.0
Via: SIP/2.0/UDP <snomip>:5060;rport;branch=z9hG4bKivuu07lcg24vgak
Max-Forwards: 70
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Contact: <sip:11@<snomip>;line=59723>
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Authorization: Digest username="11", realm="asterisk", nonce="408a5310", uri="sip:+43xxxxxxxxx@<asterisk-ip>;user=phone", response="46d9a71564327c3ba5dcea3cb597a0f7", algorithm=MD5
Content-Disposition: session
Min-SE: 90
Session-Expires: 3600
Supported: replaces,100rel,timer
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Type: application/sdp
Content-Length: 288

v=0
o=11 332833410 332833410 IN IP4 <snomip>
s=-
c=IN IP4 <snomip>
t=0 0
m=audio 50032 RTP/AVP 0 8 115 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:80
a=sendrecv
a=rtcp:50033
<------------->
--- (17 headers 15 lines) ---
Ignoring this INVITE request

<--- Transmitting (no NAT) to <snomip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Length: 0


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

<--- SIP read from UDP:<snomip>:5060 --->
INVITE sip:+43xxxxxxxxx@<asterisk-ip>;user=phone SIP/2.0
Via: SIP/2.0/UDP <snomip>:5060;rport;branch=z9hG4bKivuu07lcg24vgak
Max-Forwards: 70
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Contact: <sip:11@<snomip>;line=59723>
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Authorization: Digest username="11", realm="asterisk", nonce="408a5310", uri="sip:+43xxxxxxxxx@<asterisk-ip>;user=phone", response="46d9a71564327c3ba5dcea3cb597a0f7", algorithm=MD5
Content-Disposition: session
Min-SE: 90
Session-Expires: 3600
Supported: replaces,100rel,timer
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Type: application/sdp
Content-Length: 288

v=0
o=11 332833410 332833410 IN IP4 <snomip>
s=-
c=IN IP4 <snomip>
t=0 0
m=audio 50032 RTP/AVP 0 8 115 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:80
a=sendrecv
a=rtcp:50033
<------------->
--- (17 headers 15 lines) ---
Ignoring this INVITE request

<--- Transmitting (no NAT) to <snomip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Length: 0


<------------>
    -- Executing [+43xxxxxxxxx@from-internal:2] SIPAddHeader("SIP/11-00000078", "P-Asserted-Identity: <sip:[email protected]") in new stack

<--- SIP read from UDP:<snomip>:5060 --->
INVITE sip:+43xxxxxxxxx@<asterisk-ip>;user=phone SIP/2.0
Via: SIP/2.0/UDP <snomip>:5060;rport;branch=z9hG4bKivuu07lcg24vgak
Max-Forwards: 70
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Contact: <sip:11@<snomip>;line=59723>
Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK, UPDATE
Authorization: Digest username="11", realm="asterisk", nonce="408a5310", uri="sip:+43xxxxxxxxx@<asterisk-ip>;user=phone", response="46d9a71564327c3ba5dcea3cb597a0f7", algorithm=MD5
Content-Disposition: session
Min-SE: 90
Session-Expires: 3600
Supported: replaces,100rel,timer
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Type: application/sdp
Content-Length: 288

v=0
o=11 332833410 332833410 IN IP4 <snomip>
s=-
c=IN IP4 <snomip>
t=0 0
m=audio 50032 RTP/AVP 0 8 115 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:80
a=sendrecv
a=rtcp:50033
<------------->
--- (17 headers 15 lines) ---
Ignoring this INVITE request

<--- Transmitting (no NAT) to <snomip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Length: 0


<------------>
    -- Executing [+43xxxxxxxxx@from-internal:3] SIPAddHeader("SIP/11-00000078", "Remote-Party-ID: <sip:[email protected]") in new stack
    -- Executing [+43xxxxxxxxx@from-internal:4] Dial("SIP/11-00000078", "SIP/0043xxxxxxxxx@Easybell") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 13852
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 195.185.37.60:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK7bb36596
Max-Forwards: 70
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>
Contact: <sip:Easybell@<public-ip>:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Date: Thu, 28 Apr 2022 13:20:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: <sip:[email protected]
P-Asserted-Identity: <sip:[email protected]
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 1850281766 1850281766 IN IP4 <public-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <public-ip>
t=0 0
m=audio 13852 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/0043xxxxxxxxx@Easybell

<--- SIP read from UDP:195.185.37.60:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK7bb36596
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:195.185.37.60:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK7bb36596
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>;tag=95c37a12bff1a2c36d72bf8333176544.78550000
Call-ID: [email protected]
CSeq: 102 INVITE
P-NGCP-Auth-IP: 192.168.251.44
P-NGCP-Auth-UA: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Proxy-Authenticate: Digest realm="sip.easybell.de", nonce="YmqV3mJqlLKgvVDFV+1uw2H+mikP09bG"
Server: Sipwise NGCP Proxy 8.X
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 195.185.37.60:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK7bb36596
Max-Forwards: 70
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>;tag=95c37a12bff1a2c36d72bf8333176544.78550000
Contact: <sip:Easybell@<public-ip>:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Content-Length: 0


---
Audio is at 13852
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 195.185.37.60:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK62e0cc23
Max-Forwards: 70
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>
Contact: <sip:Easybell@<public-ip>:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Proxy-Authorization: Digest username="0049341238228", realm="sip.easybell.de", algorithm=MD5, uri="sip:[email protected]", nonce="YmqV3mJqlLKgvVDFV+1uw2H+mikP09bG", response="f648561b44f54234d45bfd7ffc80085b"
Date: Thu, 28 Apr 2022 13:20:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: <sip:[email protected]
P-Asserted-Identity: <sip:[email protected]
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 1850281766 1850281767 IN IP4 <public-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <public-ip>
t=0 0
m=audio 13852 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:195.185.37.60:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK62e0cc23
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0

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

<--- SIP read from UDP:195.185.37.60:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK62e0cc23
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>;tag=4D00259E-626A94B20008FF3A-7882A700
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS
Contact: <sip:[email protected];transport=udp>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:[email protected];transport=udp>
    -- SIP/Easybell-00000079 is ringing

<--- Transmitting (no NAT) to <snomip>:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Length: 0


<------------>
Audio is at 19818
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g726 to SDP

<--- Transmitting (no NAT) to <snomip>:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 1001524269 1001524269 IN IP4 <asterisk-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <asterisk-ip>
t=0 0
m=audio 19818 RTP/AVP 0 8 115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:195.185.37.60:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK62e0cc23
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>;tag=4D00259E-626A94B20008FF3A-7882A700
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS
Contact: <sip:[email protected];transport=udp>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:[email protected];transport=udp>
    -- SIP/Easybell-00000079 is ringing
Reliably Transmitting (no NAT) to 195.185.37.60:5060:
OPTIONS sip:sip.easybell.de SIP/2.0
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK6b2481ce
Max-Forwards: 70
From: "asterisk" <sip:Easybell@<public-ip>>;tag=as42641a5b
To: <sip:sip.easybell.de>
Contact: <sip:Easybell@<public-ip>:5060>
Call-ID: 009cc61c106b22b503819c0658327727@<public-ip>:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Date: Thu, 28 Apr 2022 13:20:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:195.185.37.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK6b2481ce
From: "asterisk" <sip:Easybell@<public-ip>>;tag=as42641a5b
To: <sip:sip.easybell.de>;tag=51414BB1-626A94B7000BE025-B68BA700
Call-ID: 009cc61c106b22b503819c0658327727@<public-ip>:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '009cc61c106b22b503819c0658327727@<public-ip>:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to <snomip>:5060:
OPTIONS sip:11@<snomip>;line=59723 SIP/2.0
Via: SIP/2.0/UDP <asterisk-ip>:5060;branch=z9hG4bK48290e4f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@<asterisk-ip>>;tag=as35b2a38c
To: <sip:11@<snomip>;line=59723>
Contact: <sip:asterisk@<asterisk-ip>:5060>
Call-ID: 46f100904163ff141baa6fd239ac72db@<asterisk-ip>:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Date: Thu, 28 Apr 2022 13:20:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (no NAT) to <snomip>:5060:
OPTIONS sip:12@<snomip>;line=10545 SIP/2.0
Via: SIP/2.0/UDP <asterisk-ip>:5060;branch=z9hG4bK4cf4b4a6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@<asterisk-ip>>;tag=as51fc3a07
To: <sip:12@<snomip>;line=10545>
Contact: <sip:asterisk@<asterisk-ip>:5060>
Call-ID: 4cdb8eee265cfaf061789c4f47ed8338@<asterisk-ip>:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Date: Thu, 28 Apr 2022 13:20:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:<snomip>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk-ip>:5060;branch=z9hG4bK48290e4f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@<asterisk-ip>>;tag=as35b2a38c
To: <sip:11@<snomip>;line=59723>
Call-ID: 46f100904163ff141baa6fd239ac72db@<asterisk-ip>:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Thu, 28 Apr 2022 13:20:55 GMT
Supported: replaces, timer
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Really destroying SIP dialog '46f100904163ff141baa6fd239ac72db@<asterisk-ip>:5060' Method: OPTIONS

<--- SIP read from UDP:<snomip>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk-ip>:5060;branch=z9hG4bK4cf4b4a6
Max-Forwards: 70
From: "asterisk" <sip:asterisk@<asterisk-ip>>;tag=as51fc3a07
To: <sip:12@<snomip>;line=10545>
Call-ID: 4cdb8eee265cfaf061789c4f47ed8338@<asterisk-ip>:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Thu, 28 Apr 2022 13:20:55 GMT
Supported: replaces, timer
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Really destroying SIP dialog '4cdb8eee265cfaf061789c4f47ed8338@<asterisk-ip>:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to <snomip>:5060:
OPTIONS sip:22@<snomip>;line=10710 SIP/2.0
Via: SIP/2.0/UDP <asterisk-ip>:5060;branch=z9hG4bK0dffb2dd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@<asterisk-ip>>;tag=as0cd0dc39
To: <sip:22@<snomip>;line=10710>
Contact: <sip:asterisk@<asterisk-ip>:5060>
Call-ID: 56223b3a68b9ed6d546b0a823f02b221@<asterisk-ip>:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Date: Thu, 28 Apr 2022 13:20:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:<snomip>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk-ip>:5060;branch=z9hG4bK0dffb2dd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@<asterisk-ip>>;tag=as0cd0dc39
To: <sip:22@<snomip>;line=10710>
Call-ID: 56223b3a68b9ed6d546b0a823f02b221@<asterisk-ip>:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Date: Thu, 28 Apr 2022 13:20:55 GMT
Supported: replaces, timer
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Really destroying SIP dialog '56223b3a68b9ed6d546b0a823f02b221@<asterisk-ip>:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 195.185.37.60:5060:
OPTIONS sip:sip.easybell.de SIP/2.0
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK1eb3786b
Max-Forwards: 70
From: "asterisk" <sip:Easybell@<public-ip>>;tag=as6218cff7
To: <sip:sip.easybell.de>
Contact: <sip:Easybell@<public-ip>:5060>
Call-ID: 1a7db3435a9bfde8612de8a55278e97b@<public-ip>:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Date: Thu, 28 Apr 2022 13:20:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:195.185.37.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK1eb3786b
From: "asterisk" <sip:Easybell@<public-ip>>;tag=as6218cff7
To: <sip:sip.easybell.de>;tag=778FA80E-626A94B800010000-B6ABC700
Call-ID: 1a7db3435a9bfde8612de8a55278e97b@<public-ip>:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1a7db3435a9bfde8612de8a55278e97b@<public-ip>:5060' Method: OPTIONS

<--- SIP read from UDP:195.185.37.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK62e0cc23
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>;tag=4D00259E-626A94B20008FF3A-7882A700
Call-ID: [email protected]
CSeq: 103 INVITE
Supported: histinfo, x-diversion
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS
Contact: <sip:[email protected];transport=udp>
Content-Type: application/sdp
Content-Length: 222

v=0
o=- 3911378507 1861284342 IN IP4 195.185.37.60
s=-
c=IN IP4 195.185.37.60
t=0 0
m=audio 36420 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=direction:both
<------------->
--- (11 headers 11 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 - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x74303350 -- Strict RTP learning after remote address set to: 195.185.37.60:36420
Peer audio RTP is at port 195.185.37.60:36420
sip_route_dump: route/path hop: <sip:[email protected];transport=udp>
set_destination: Parsing <sip:[email protected];transport=udp> for address/port to send to
set_destination: set destination to 195.185.37.60:5060
Transmitting (no NAT) to 195.185.37.60:5060:
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK054d4bbc
Max-Forwards: 70
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>;tag=4D00259E-626A94B20008FF3A-7882A700
Contact: <sip:Easybell@<public-ip>:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Content-Length: 0


---
    -- SIP/Easybell-00000079 answered SIP/11-00000078
Audio is at 19818
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g726 to SDP

<--- Reliably Transmitting (no NAT) to <snomip>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 1001524269 1001524269 IN IP4 <asterisk-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <asterisk-ip>
t=0 0
m=audio 19818 RTP/AVP 0 8 115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/Easybell-00000079 joined 'simple_bridge' basic-bridge <bb74c7ea-09ce-43d9-ac4e-9ef52a7c197a>
    -- Channel SIP/11-00000078 joined 'simple_bridge' basic-bridge <bb74c7ea-09ce-43d9-ac4e-9ef52a7c197a>
Retransmitting #1 (no NAT) to <snomip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 1001524269 1001524269 IN IP4 <asterisk-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <asterisk-ip>
t=0 0
m=audio 19818 RTP/AVP 0 8 115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=maxptime:150
a=sendrecv

---
Retransmitting #2 (no NAT) to <snomip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 1001524269 1001524269 IN IP4 <asterisk-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <asterisk-ip>
t=0 0
m=audio 19818 RTP/AVP 0 8 115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=maxptime:150
a=sendrecv

---
Retransmitting #3 (no NAT) to <snomip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 1001524269 1001524269 IN IP4 <asterisk-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <asterisk-ip>
t=0 0
m=audio 19818 RTP/AVP 0 8 115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=maxptime:150
a=sendrecv

---
Retransmitting #4 (no NAT) to <snomip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 1001524269 1001524269 IN IP4 <asterisk-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <asterisk-ip>
t=0 0
m=audio 19818 RTP/AVP 0 8 115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=maxptime:150
a=sendrecv

---
Retransmitting #5 (no NAT) to <snomip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 1001524269 1001524269 IN IP4 <asterisk-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <asterisk-ip>
t=0 0
m=audio 19818 RTP/AVP 0 8 115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=maxptime:150
a=sendrecv

---
Retransmitting #6 (no NAT) to <snomip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <snomip>:5060;branch=z9hG4bKivuu07lcg24vgak;received=<snomip>;rport=5060
From: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
To: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 41673 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+43xxxxxxxxx@<asterisk-ip>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 253

v=0
o=root 1001524269 1001524269 IN IP4 <asterisk-ip>
s=Asterisk PBX 13.14.1~dfsg-2+deb9u3
c=IN IP4 <asterisk-ip>
t=0 0
m=audio 19818 RTP/AVP 0 8 115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:115 G726-32/8000
a=maxptime:150
a=sendrecv

---
[Apr 28 15:21:06] WARNING[806]: chan_sip.c:4071 retrans_pkt: Retransmission timeout reached on transmission .w5c8kw7kdhrd833f@<asterisk-ip> for seqno 41673 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[Apr 28 15:21:06] WARNING[806]: chan_sip.c:4095 retrans_pkt: Hanging up call .w5c8kw7kdhrd833f@<asterisk-ip> - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Channel SIP/11-00000078 left 'simple_bridge' basic-bridge <bb74c7ea-09ce-43d9-ac4e-9ef52a7c197a>
  == Spawn extension (from-internal, +43xxxxxxxxx, 4) exited non-zero on 'SIP/11-00000078'
    -- Channel SIP/Easybell-00000079 left 'simple_bridge' basic-bridge <bb74c7ea-09ce-43d9-ac4e-9ef52a7c197a>
Scheduling destruction of SIP dialog '.w5c8kw7kdhrd833f@<asterisk-ip>' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:11@<snomip>;line=59723> for address/port to send to
set_destination: set destination to <snomip>:5060
set_destination: Parsing <sip:[email protected];transport=udp> for address/port to send to
Reliably Transmitting (no NAT) to <snomip>:5060:
BYE sip:11@<snomip>;line=59723 SIP/2.0
Via: SIP/2.0/UDP <asterisk-ip>:5060;branch=z9hG4bK6c9642b0;rport
Max-Forwards: 70
From: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
To: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Proxy-Authorization: Digest username="11", realm="asterisk", algorithm=MD5, uri="sip:<asterisk-ip>", nonce="408a5310", response="fda1009519d755a626d74fc84112aa27"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---
set_destination: set destination to 195.185.37.60:5060
Reliably Transmitting (no NAT) to 195.185.37.60:5060:
BYE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK0766b93e
Max-Forwards: 70
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>;tag=4D00259E-626A94B20008FF3A-7882A700
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Proxy-Authorization: Digest username="0049341238228", realm="sip.easybell.de", algorithm=MD5, uri="sip:[email protected]", nonce="YmqV3mJqlLKgvVDFV+1uw2H+mikP09bG", response="61c18140a05c91b1108fc36d163c31b4"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---

<--- SIP read from UDP:<snomip>:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP <asterisk-ip>:5060;branch=z9hG4bK6c9642b0;rport=5060
Max-Forwards: 70
From: <sip:+43xxxxxxxxx@<asterisk-ip>;user=phone>;tag=as152c410b
To: "<name>" <sip:11@<asterisk-ip>>;tag=ckbx4dco7k
Call-ID: .w5c8kw7kdhrd833f@<asterisk-ip>
CSeq: 102 BYE
Proxy-Authorization: Digest realm="asterisk", nonce="408a5310", algorithm=MD5, username="11", response="fda1009519d755a626d74fc84112aa27", uri="sip:<asterisk-ip>"
User-Agent: snomM300/05.20.0001 (MAC=000413621A19; SER= 00000; HW=1)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '.w5c8kw7kdhrd833f@<asterisk-ip>' Method: INVITE

<--- SIP read from UDP:195.185.37.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK0766b93e
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>;tag=4D00259E-626A94B20008FF3A-7882A700
Call-ID: [email protected]
CSeq: 104 BYE
Content-Length: 0

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

koyaanisqatsi · 28 Apr 2022

Moinsen

1. Eine vernünftige Codecreihenfolge sollte so aussehen...
allow=!all,alaw,ulaw <--<< Bei Problemen mit Codec bedeutet dies: Nur G.711 - PCMA zuerst anbieten
Nach dem Motto: Der Gewünschte kommt zuerst und hat die höchste Priorität.
Bei mir wäre g722 der Gewünschte und damit...
allow=!all,g722,alaw,ulaw

2. Die NAT Erkennung

Rich (BBCode):

Network Settings:
---------------------------
  SIP address remapping:  Enabled using externaddr
  Externhost:             <none>
  Externaddr:             77.24.110.145:0
  Externrefresh:          10
  Localnet:               192.168.63.0/255.255.255.0

Das sollte nach meinen Verständnis so aussehen...

Rich (BBCode):

Network Settings:
---------------------------
  SIP address remapping:  Enabled using externaddr
  Externhost:             <none>
  Externaddr:             77.24.110.145:5060 ; Mit Portangabe
  Externrefresh:          120 ; 2 Minuten (Ich hab hier 180 also 3 Minuten)
  Localnet:               192.168.0.0/255.255.255.0 ; LAN/WLAN

Auch ist es von Vorteil in res_stun_monitor.conf einen STUN server einzutragen.
STUN muss dabei gar nicht genutzt werden.
Es dient aber bei chan_sip.so und chan_iax.so dass die Registrierungen erneuert werden wenn sich deine IP ändert.

Beispiel res_stun_monitor.conf

Rich (BBCode):

;
; Configuration file for the res_stun_monitor module
;
; The res_stun_monitor module sends STUN requests to a configured STUN server
; periodically.  If the monitor detects a change in the external IP address or port
; provided by the STUN server an event is sent out internally within Asterisk
; to alert all listeners to that event of the change.

; The current default listeners for the network change event include chan_sip
; and chan_iax.  Both of these channel drivers by default react to this event
; by renewing all outbound registrations.  This allows the endpoints Asterisk
; is registering with to become aware of the address change and know the new
; location.
;
[general]
;
; ---- STUN Server configuration ---
;  Setting the 'stunaddr' option to a valid address enables the STUN monitor.
;
stunaddr = stun.1und1.de        ; Address of the STUN server to query.
                                ; Valid form:
                                ;   [(hostname | IP-address) [':' port]]
                                ; The port defaults to the standard STUN port (3478).
                                ; Set to an empty value to disable STUN monitoring.
                                ;   Default is disabled.
stunrefresh = 30                ; Number of seconds between STUN refreshes.
                                ;   Default is 30.

Dann kannste auch mit oder in der *-Konsole deine öffentliche IP checken mit...

Code:

asterisk -x 'stun show status'
Hostname                  Port  Period  Retries  Status  ExternAddr       ExternPort
stun.1und1.de             3478  30      3        OK      46.142.33.143    33495

...eben auch wenn STUN/ICE gar nicht genutzt wird.

Lomas · 28 Apr 2022

Danke für Deine Antwort. Den Port und die STUN-Einstellungen werde ich ergänzen. Warum kann für localnet nicht mein lokales Netzwerk angegeben werden, wenn ich dann doch wieder eine Class C-Maske angebe?

Was mich beschäftigt, ist, dass das Problem auch bei internen Anrufen auftritt. Also: Ich rufe an, mein Telefon legt auf, und erst dann sehe ich die Aktivität im CLI und es klingelt beim Angerufenen. Die obigen Änderungen haben doch damit nichts zu tun, oder irre ich mich da?

koyaanisqatsi · 28 Apr 2022

Es ist ein Netz und keine Adresse.
Deswegen so viele Nullen wie möglich.
Also alles was sich unter 192.168.*.* tummelt ist: NAT (LAN/WLAN)

Lomas schrieb:
Was mich beschäftigt, ist, dass das Problem auch bei internen Anrufen auftritt. Also: Ich rufe an, mein Telefon legt auf, und erst dann sehe ich die Aktivität im CLI und es klingelt beim Angerufenen. Die obigen Änderungen haben doch damit nichts zu tun, oder irre ich mich da?

Du meinst...

Rich (BBCode):

set_destination: set destination to 195.185.37.60:5060
Reliably Transmitting (no NAT) to 195.185.37.60:5060:
BYE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP <public-ip>:5060;branch=z9hG4bK0766b93e
Max-Forwards: 70
From: "<name>" <sip:[email protected]>;tag=as613210c0
To: <sip:[email protected]>;tag=4D00259E-626A94B20008FF3A-7882A700
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Proxy-Authorization: Digest username="00493XXXXXX28", realm="sip.easybell.de", algorithm=MD5, uri="sip:[email protected]", nonce="YmqV3mJqlLKgvVDFV+1uw2H+mikP09bG", response="61c18140a05c91b1108fc36d163c31b4"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

Das wird sechsmal versucht und deutet daraufhin, dass die NAT Erkennung schlichtweg versagt hat.

IEEE · 29 Apr 2022

Ich mach bei jedem Asterisk immer standardmäßig die Zeilen

Code:

localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0    ; Also RFC1918
localnet=172.16.0.0/12         ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network

rein. Hab ich mir mal als Tipp (ich glaub sogar von hier) geben lassen und es bewährt sich, weil damit alle als privat definierten Netze automatisch als localnet definiert sind.

Ansonsten find ich beim aktuellen Fall sehr eigenartig dass der Threadstarter sagt, dass das Problem auch bei rein internen Calls auftritt...

Lomas · 29 Apr 2022

Genau so ist es. Die Debug-Ausgaben zeigen einen Anruf bei einer externen Rufnummer. Einen Mitschnitt von einem internen Anruf habe ich nicht. Wenn ich beispielsweise vom Apparat 22 die interne Rufnummer 11 (oder eine andere Nr.) wähle, passiert im CLI gar nichts. Dann legt mein Telefon auf und erst anschließend sehe ich die Ausgaben im CLi und Apparat 11 klingelt. Manchmal funktioniert es ja auch. Normalerweise sind die erscheinen die Ausgaben sofort, und zwar schon bevor es beim Angerufenen klingelt. Das Verhalten ist bei internen und externen Anrufen gleich, und machmal funktioniert es und oft auch nicht. Zumindest bei den internen Anrufen sollte doch NAT keine Rolle spielen.

Lomas · 29 Apr 2022

Ich habe das jetzt noch einmal mit PhonerLite probiert. Das funktioniert, allerdings dauert es fast 20s, bevor der Ruf abgeht. D.h. es ist 20s lang Ruhe, dann sehe ich die entsprechenden Ausgaben im CLI, der Rufton im PhonerLite ist zu hören und es klingelt beim Angerufenen. Der Unterschied zum Snom ist, dass PhonerLite nicht in der Zwischenzeit auflegt.

IEEE · 29 Apr 2022

Ich denke mittlerweile nicht mehr, dass das irgendwas mit NAT Problemen zu tun hat.
Letztendlich zeigt sich das Verhalten bei Deinem PhonerLite Test ident mit den Problemen auf den regulären Telefonen - Phonerlite scheint halt nur geduldiger zu sein und wartet die 20 Sekunden, wo andere Geräte aufgeben.

Im nächsten Schritt müsste wohl geklärt werden warum die SIP Pakete manchmal so stark verzögert beim Asterisk aufschlagen. Aus der Erfahrung heraus würde ich da mal ziemlich Low Level beginnen einzugrenzen. Letztendlich kann selbst ein verschnupfter Switch sowas auslösen. Also je nach Möglichkeit die Asterisk Maschine mal an einen anderen Switchport / andere Dose hängen, Patchkabel tauschen etc.

Lomas · 1 Mai 2022

Ich werde wohl zunächst einen Asterisk auf einer anderen Maschine aufsetzen, denn dass die Pakete 20s im lokalen Netz umher irren und dann doch ankommen, kann ich mir nur schwer vorstellen. Ich werde berichten.

IEEE · 1 Mai 2022

Wie gesagt, ich kenn sowas schon auch bei Netzwerkproblemen auf den untersten Schichten. Einmal hat mir jemand bei einem Switch einen Loop gebaut, da gabs dann auch sehr starke Verzögerungen. Aber ich erinnere mich leider nicht mehr genau.

Interessanterweise hat im asterisk Forum grad einer ein Problem, bei dem ich mal vorsichtig sagen würde dass es Deinem gar nicht so unähnlich ist. Kam aber noch nichts raus dabei.

The call does not come for a long time and crashes

Please help me, I’ve been struggling for a week now, I can’t solve the problem. There is: centos 7 Asterisk 18.11.1 When I call an internal number, I see in the log that the request goes immediately, instantly, and then some long pause of 20-30 seconds, sometimes more, up to a minute, and...

community.asterisk.org

Lomas · 1 Mai 2022

Werde ich untersuchen. Allerdings wurde am Netzwerk nicht geändert. Und ich habe das Gefühl, dass es schlimmer wird. Als die Probleme begannen, fubnktionierte es nach einem "core reload" zumindest kurzzeitig wieder.

gigauser · 12 Jun 2022

Ist das Problem schon gelöst?

Lomas · 13 Jun 2022

Ist das Problem schon gelöst?

Ich habe den Asterisk auf einer neuen Maschine mit einer aktuellen Version (war vorher ein Raspberry mit Asterisk 13) und diesmal mit pjsip statt chan_sip aufgesetzt. Seitdem funktioniert es. Eine weitere Ursachenforschung habe ich nicht unternommen.

Verzögerung bei ausgehenden Anrufen

Neuer User

IPPF-Urgestein

Neuer User

Neuer User

IPPF-Urgestein

Neuer User

IPPF-Urgestein

Mitglied

Neuer User

Neuer User

Mitglied

Neuer User

Mitglied

Neuer User

Neuer User

Neuer User

Statistik des Forums