[Problem] Fehlermeldung Anmeldung: No registered publish handler for event presence

jeschero · 15 Mai 2016

Hallo alle zusammen,

und zwar habe ich ein kleines Problem.
Ich bin gerade dabei Asterisk einzurichten.

Telefonieren funktioniert jetzt, siehe Edit4.
und zwar kann ich nach einem restart vom Asterisk sauber raustelefonieren, wenn ich rund 1 Minuten/ 40 Sekunden warten bekommen ich bei Zioper die folgende Fehlermeldung: SIP 403 Forbidden Dienstkennung nicht autorisiert.
Errlog von Asterisk habe ich angehängt
Ich hoffe mir kann jemand helfen.

Config: Datei von filehorst.de laden
Siehe Edit 5.

Port Forward Rules für Asterisk in Pfsense:

Edit 1:
Von

auf

Code:

 geändert.
Danke für den Hinweis.

Edit 2:
Durch rentier-s Hilfe wurde die Config angepasst und ich kann mich jetzt Erfolgreich anmelden.

Leider ist kein raustelefonieren möglich und ankommen funktionieren auch nicht.

Edit 3:
Raustelefonieren funktioniert.
Ankommende Gespräche werden nicht angenommen/durchgereicht.
Das Handy legt einfach auf.

Edit 4:
Raus und reintelefonieren funktioniert jetzt prima.
Der identify-Block für Telekom_in hat gefehlt.

Jetzt habe ich das Problem, das ich noch rund 1 Minuten/ 40 Sekunden beim Raustelefonieren eine Fehlermeldung bekomm.
SIP 403 Forbidden 
Dienstkennung nicht autorisiert.

Errorlog von Asterisk ist oben

Edit 5:
Config: [url=http://filehorst.de/d/beorCmCu]Datei von filehorst.de laden[/url]

Nach einem Neustart von Asterisk kann rein- und raus telefoniert werden, aber es kommt an beiden Seitenkein Ton an.
Nach wenigen Minuten kann nicht mehr rein und raus telefoniert werden.

Errorlog:
[url=http://filehorst.de/d/baDCzyou]Datei von filehorst.de laden[/url]
(Ist zu groß für den Post und der File-Manager funktioniert bei  mir nicht)

koyaanisqatsi · 15 Mai 2016

Moins

Bitte den ersten Beitrag editieren und die [ QUOTE ] Tags in [ CODE ] ändern.
Das steigert die Lesbarkeit enorm und stellt es nicht kursiv dar.

PS: Willkommen im Forum

rentier-s · 17 Mai 2016

Das Telefon versucht sich als user anzumelden, der Endpoint heißt aber mein-telefon.

jeschero · 17 Mai 2016

OK,
was muss ich den bei Zioper angeben um mich anzumelden?

Ich habe jetzt das eingetragen und es funktioniert nicht:
Kontotyp: SIP
Benutzer / Benutzer@Host: user
Password: erdshn235
Domain / Outbound-Proxy 172.28.131.216

Bsz. Wie muss ich die Config anpassen?

Wäre dir für deine Hilfe echt dankbar

rentier-s · 17 Mai 2016

was muss ich den bei Zioper angeben um mich anzumelden?

Benutzername (Anmeldename): mein-telefon
Auth-User: user

Bsz. Wie muss ich die Config anpassen?

[mein-telefon] muss [user] heißen, dann gehts mit Benutzername user.

https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships schrieb:
However, in some cases, (endpoint and aor types) the section name has a relationship to its function. In the case of endpoint and aor their names must match the user portion of the SIP URI in the "To" header for inbound SIP requests.

Übrigens solltest Du noch das expire beim T-Com Peer auf 3600 stellen, sonst lehnen deren Server die Registrierung ab.

jeschero · 17 Mai 2016

Danke rentier-s für deine Hilfe.

Anmelden funktioniert jetzt.

Wegen dass:

Übrigens solltest Du noch das expire beim T-Com Peer auf 3600 stellen, sonst lehnen deren Server die Registrierung ab.

Ich habe jetzt expiration auf 3600 geändert.

Kannst du mir sagen, was ich gegen die neue Fehlermeldung machen kann?
Bekomme jetzt die Fehlermeldung: WARNING[3327] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfo

Post habe ich oben aktualisiert.

Schon mal Danke für deine Hilfe.

Edit 1:
Raustelefonieren funktioniert.
Nur ankommende Anrufen nicht. Errorlog habe ich oben Aktualisiert

rentier-s · 18 Mai 2016

Für Ankommend brauchst Du vermutlich ein identify, wobei ich ehrlich gestehen muss, so genau habe ich das selbst noch nicht kapiert. PJSIP läuft bei mir auf dem Pi nicht, deshalb habe ich da noch keine praktische Erfahrung. T-Com ist wegen der Verwendung von Load Balancern sowieso problematisch zu konfigurieren, ich nutze dafür den default Context.

Ist übrigens nicht so optimal den Eingangspost komplett über den Haufen zu werfen, so kann niemand mehr das urpsüngliche Problem und die Beiträge dazu nachvollziehen. Nächstes Mal bitte einfach hinten dran hängen, oder gar ein neues Thema aufmachen.

jeschero · 18 Mai 2016

Danke für deinen Hinweis, werde ich in zukunft berücksichtigen.

Ich bin jetzt aber schonmal ein Schritt weiter.
Telefonieren funktioniert, zuminidstens für rund 40 Sekunden, dann bekomme ich eine neue Fehlermeldung, siehe Post oben:
SIP 403 Forbidden
Dienstkennung nicht autorisiert
und:

Code:

...
[May 18 16:31:39] WARNING[3125] res_pjsip_pubsub.c: No registered publish handler for event presence
[May 18 16:31:39] VERBOSE[3125] res_pjsip_logger.c: <--- Transmitting  SIP response (348 bytes) to UDP:172.28.131.50:34265 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 172.28.131.50:34265;rport=34265;received=172.28.131.50;branch=z9hG4bK-524287-1---9b75873d4d003b07
Call-ID: IrN8_Jepa3QrE-n4xb3Feg..
From: <sip:[email protected]>;tag=d9785d23
To: <sip:[email protected]>;tag=z9hG4bK-524287-1---9b75873d4d003b07
CSeq: 2 PUBLISH
Server: Asterisk
Content-Length:  0
...

rentier-s · 19 Mai 2016

Das Bad Event kommt davon, weil Asterisk das PUBLISH nicht versteht, das sollte aber eigentlich zu keinen Gesprächsabbrüchen führen.

Asterisk gibt im INVITE seine lokale IP-Adresse an, deshalb antwortet die T-Com mit Forbidden.
https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT

jeschero · 19 Mai 2016

Ich habe mir dein Link angeschaut. dort steht, dass ich bei NAT folgendes nutzen soll:
external_media_address

external_signaling_address

Da steht ich soll IP-Adressen angeben, meine ist aber Dynamisch und dynDNS-Namem nimmt er schon an, ich kann dann aber nicht mehr raustelefonieren.
Und jeden morgen um zwei die IP-Adrsse anzupassen, bringt nicht wirklich spass

Also fällt das wohl weg, oder kennt Ihr eine andere Möglichkeit?

Außerdem habe ich immer noch das Problem, dass ich nach rund 40 Sekunden/1 Minute nicht mehr raus und reintelefonieren kann.
Beim Raustelefonieren kommt die Fehlermeldung von oben und beim Reintelefonieren wird nichts im Errorlog geschrieben.

Edit 1:
Als versuch wurde "icesupport=yes" inder pjsip.conf hinzugefügt und unter der rtp.conf "stunaddr=stun.t-online.de".
Die Verbindung hällt jetzt rund 5 min.

Wenn ich danach versuche zu rauszutelefonieren gibt es jetzt keine Fehlermeldung in Zoiper, sondern nur die Warnung:
[May 19 17:25:10] WARNING[1114] res_pjsip_pubsub.c: No registered publish handler for event presence
[May 19 17:25:10] WARNING[1114] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfo
[May 19 17:25:15] WARNING[1113] res_pjsip_pubsub.c: No registered publish handler for event presence
[May 19 17:25:15] WARNING[1113] res_pjsip_pubsub.c: No registered subscribe handler for event presence.winfo

rentier-s · 20 Mai 2016

PJSIP kann STUN, frag mich aber bitte nicht, wie man das in chan_pjsip konfigurieren muss. Es wundert mich nur gerade, dass man keine Hostnamen verwenden kann, das wäre ja ein Rückschritt.

Als Notlösung könntest Du execincludes in der asterisk.conf einschalten, ein kleines Skript bauen das die externe IP Adresse über irgendeinen Dienst abfrägt und die beiden Parameter mit zB. echo ausgibt. Das wird mit #exec Pfad/zum/Skript in die .conf eingebunden und mit jedem reload ausgeführt.

Fakt ist, das T-Com System ist absolut nicht NAT tolerant, so lange Asterisk seine externe Adresse nicht kennt wird das Probleme geben.

rmh · 21 Mai 2016

@jeschero, Hast du in pfsense auch eine outbound nat regel für deinen sip und rtp traffic hinterlegt? Das bewirkt Wunder.

jeschero · 23 Mai 2016

@rentier-s, ich habe versuchshalbe meine aktuelle ip in der pjsip config eingetragen:
external_media_address=xxx.xxx.xxx.xxx
external_signaling_address=xxx.xxx.xxx.xxx

Leider hat dies nicht zum Erfolg geführt. ich bekomme immer noch bei Zoiper den Fehler 403 und die Fehlermeldung wie vom letzten Post.

@rmh, ich habe mal die Regel hinzugefügt, gebracht hat das leider nichts

.

rentier-s · 24 Mai 2016

Zeig bitte noch mal ein aktuelles Log (SIP Debug) eines REGISTER und des INVITE beim raus rufen. Die T-Com verlangt eine erfolgreiche Registrierung für ausgehende Gespräche von derselben IP-Adresse und Port, sonst gibts ein 403.

jeschero · 24 Mai 2016

Log vom raus rufen. Nach https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information eingestellt

xxxx ist die externe Nummer
yyyy ist die interne Nummer

Start von Asterisk

Code:

[May 24 15:29:09] VERBOSE[9906] res_pjsip_logger.c: <--- Transmitting SIP request (564 bytes) to UDP:217.0.23.100:5060 --->
REGISTER sip:tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 93.198.207.225:5060;rport;branch=z9hG4bKPj47b08f82-9296-4ff3-8451-9599ea952dbf
From: <sip:[email protected]>;tag=941727dc-fd74-4f20-b017-708b60b902e4
To: <sip:[email protected]>
Call-ID: 1954dd6e-0ae7-422d-933d-f6aa35d4f660
CSeq: 64057 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 480
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 24 15:29:09] VERBOSE[9906] res_pjsip_logger.c: <--- Received SIP response (608 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 401 Unauthorized 010330345
Via: SIP/2.0/UDP 93.198.207.225:5060;received=84.130.196.114;rport=37489;branch=z9hG4bKPj47b08f82-9296-4ff3-8451-9599ea952dbf
To: <sip:[email protected]>;tag=h7g4Esbg_34ed7b21079fe434d0c1ef5d5b1219bf
From: <sip:[email protected]>;tag=941727dc-fd74-4f20-b017-708b60b902e4
Call-ID: 1954dd6e-0ae7-422d-933d-f6aa35d4f660
CSeq: 64057 REGISTER
Service-Route: <sip:217.0.23.100:5060;transport=udp;lr>
WWW-Authenticate: Digest realm="tel.t-online.de",nonce="F36098893057445700000",stale=true,algorithm=MD5,qop="auth"
Content-Length: 0


[May 24 15:29:09] VERBOSE[9907] res_pjsip_logger.c: <--- Transmitting SIP request (834 bytes) to UDP:217.0.23.100:5060 --->
REGISTER sip:tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 93.198.207.225:5060;rport;branch=z9hG4bKPjefa3a380-0de8-4880-b862-8a975d08f978
From: <sip:[email protected]>;tag=941727dc-fd74-4f20-b017-708b60b902e4
To: <sip:[email protected]>
Call-ID: 1954dd6e-0ae7-422d-933d-f6aa35d4f660
CSeq: 64058 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 480
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Max-Forwards: 70
User-Agent: Asterisk
Authorization: Digest username="0yyyy", realm="tel.t-online.de", nonce="F3609889305744570000000028A38473", uri="sip:tel.t-online.de", response="867dff5758932c4335db737", algorithm=MD5, cnonce="04507125-3e16-4618-a27c", qop=auth, nc=00000001
Content-Length:  0


[May 24 15:29:09] VERBOSE[9906] res_pjsip_logger.c: <--- Received SIP response (747 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 93.198.207.225:5060;received=84.130.196.114;rport=37489;branch=z9hG4bKPjefa3a380-0de8-4880-b862-8a975d08f978
To: <sip:[email protected]>;tag=h7g4Esbg_34ed7b2107a01a2bd0c1ef5d5b41205b
From: <sip:[email protected]>;tag=941727dc-fd74-4f20-b017-708b60b902e4
Call-ID: 1954dd6e-0ae7-422d-933d-f6aa35d4f660
CSeq: 64058 REGISTER
Contact: <sip:[email protected]:5060>;expires=480
P-Associated-Uri: <sip:[email protected]>
P-Associated-Uri: <tel:+49yyyy>
Service-Route: <sip:217.0.23.100:5060;transport=udp;lr>
Content-Length: 0
Authentication-Info: qop=auth,rspauth="bf93a4c5e2adb1f61b0794d55d86f18b",cnonce="04507125-3e16-4618-a27c-5db07b5b3f29",nc=00000001

Log, sobald es nicht mehr funktioniert

Code:

[May 24 15:21:20] VERBOSE[8923] res_pjsip_logger.c: <--- Transmitting SIP request (1016 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 93.198.207.225:5060;rport;branch=z9hG4bKPj3e1a78ca-f935-437c-b085-9f6cc13f7ce7
From: <sip:[email protected]>;tag=a28cb29e-bda4-4d5d-a415-e9df09328405
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 577592d1-3d29-4397-b732-d8575d583301
CSeq: 13215 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   343

v=0
o=- 2102104386 2102104386 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 93.198.207.225
t=0 0
m=audio 30216 RTP/AVP 8 111 9 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 24 15:21:20] VERBOSE[8923] res_pjsip_logger.c: <--- Received SIP response (406 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 93.198.207.225:5060;received=84.130.196.114;rport=17431;branch=z9hG4bKPj3e1a78ca-f935-437c-b085-9f6cc13f7ce7
To: <sip:[email protected]>;tag=h7g4Esbg_b5s3kpgjelyhjmrqbyrhpwk5hdoxlk88
From: <sip:[email protected]>;tag=a28cb29e-bda4-4d5d-a415-e9df09328405
Call-ID: 577592d1-3d29-4397-b732-d8575d583301
CSeq: 13215 INVITE
Content-Length: 0

rentier-s · 25 Mai 2016

Asterisk gibt als Audio-Ziel seine lokale IP-Adresse an, das mag die T-Com nicht.

o=- 2102104386 2102104386 IN IP4 172.28.131.216

Warum er das trotz external_signaling_address macht :noidea:

Außerdem ist das noch etwas seltsam:

SIP/2.0/UDP 93.198.207.225:5060;received=84.130.196.114;rport=37489

Die Adressen sollten eigentlich übereinstimmen :gruebel:

jeschero · 25 Mai 2016

Ein Fehler konnte ich finden, der DDNS hat sich nicht aktualisiert.
Jetzt geht er aber wieder.

Meine Fehlermeldung über Zoiper ist jetzt : SIP 488 - Not Acceptable Here

LOG:

Code:

[May 25 15:25:09] VERBOSE[12379] res_pjsip_logger.c: <--- Transmitting SIP request (564 bytes) to UDP:217.0.23.100:5060 --->
REGISTER sip:tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 84.130.999.999:5060;rport;branch=z9hG4bKPj2c415980-916c-455b-8533-b956dadebf7e
From: <sip:[email protected]>;tag=8b23c7ce-a3c3-40b5-b3c0-da93b5253429
To: <sip:[email protected]>
Call-ID: b7cb1ff1-b575-48fc-bfd6-f1e1a75872d7
CSeq: 47707 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 480
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0

[May 25 15:25:09] VERBOSE[12379] res_pjsip_logger.c: <--- Received SIP response (608 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 401 Unauthorized 010330345
Via: SIP/2.0/UDP 84.130.999.999:5060;received=84.130.999.999;rport=43786;branch=z9hG4bKPj2c415980-916c-455b-8533-b956dadebf7e
To: <sip:[email protected]>;tag=h7g4Esbg_2a51588c07bb8be060c1f4800d15e57e
From: <sip:[email protected]>;tag=8b23c7ce-a3c3-40b5-b3c0-da93b5253429
Call-ID: b7cb1ff1-b575-48fc-bfd6-f1e1a75872d7
CSeq: 47707 REGISTER
Service-Route: <sip:217.0.23.100:5060;transport=udp;lr>
WWW-Authenticate: Digest realm="tel.t-online.de",nonce="65A54230C0A745570000000058890240",stale=true,algorithm=MD5,qop="auth"
Content-Length: 0

[May 25 15:25:09] VERBOSE[12380] res_pjsip_logger.c: <--- Transmitting SIP request (834 bytes) to UDP:217.0.23.100:5060 --->
REGISTER sip:tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 84.130.999.999:5060;rport;branch=z9hG4bKPj8635f5f7-47e2-4303-87d0-fb0ca626dcae
From: <sip:[email protected]>;tag=8b23c7ce-a3c3-40b5-b3c0-da93b5253429
To: <sip:[email protected]>
Call-ID: b7cb1ff1-b575-48fc-bfd6-f1e1a75872d7
CSeq: 47708 REGISTER
Contact: <sip:[email protected]:5060>
Expires: 480
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Max-Forwards: 70
User-Agent: Asterisk
Authorization: Digest username="0yyyy", realm="tel.t-online.de", nonce="65A54230C0A745570000000058890240", uri="sip:tel.t-online.de", response="a64eef8feff339fb799ade07643dec89", algorithm=MD5, cnonce="a168c7e1-cb3c-478c-9069-dd4b1d3cd56d", qop=auth, nc=00000001
Content-Length:  0

[May 25 15:25:09] VERBOSE[12379] res_pjsip_logger.c: <--- Received SIP response (747 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 84.130.999.999:5060;received=84.130.999.999;rport=43786;branch=z9hG4bKPj8635f5f7-47e2-4303-87d0-fb0ca626dcae
To: <sip:[email protected]>;tag=h7g4Esbg_2a51588c07bb6ad1c0c1f4800d507556
From: <sip:[email protected]>;tag=8b23c7ce-a3c3-40b5-b3c0-da93b5253429
Call-ID: b7cb1ff1-b575-48fc-bfd6-f1e1a75872d7
CSeq: 47708 REGISTER
Contact: <sip:[email protected]:5060>;expires=480
P-Associated-Uri: <sip:[email protected]>
P-Associated-Uri: <tel:+49yyyy>
Service-Route: <sip:217.0.23.100:5060;transport=udp;lr>
Content-Length: 0
Authentication-Info: qop=auth,rspauth="aa3768be84a30af3c30dd34c3bcd5a8e",cnonce="a168c7e1-cb3c-478c-9069-dd4b1d3cd56d",nc=00000001

Code:

[May 25 15:26:15] VERBOSE[12379] res_pjsip_logger.c: <--- Received SIP request (838 bytes) from UDP:172.28.131.52:32368 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.28.131.52:32368;branch=z9hG4bK-524287-1---88cb65a5723bd495
Max-Forwards: 70
Contact: <sip:[email protected]:32368;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=c6742c10
Call-ID: rDpDbJUXg_dnI87iBuyqFA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 167

v=0
o=Z 0 0 IN IP4 172.28.131.52
s=Z
c=IN IP4 172.28.131.52
t=0 0
m=audio 8000 RTP/AVP 3 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv


[May 25 15:26:15] VERBOSE[12402] res_pjsip_logger.c: <--- Transmitting SIP response (505 bytes) to UDP:172.28.131.52:32368 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.28.131.52:32368;rport=32368;received=172.28.131.52;branch=z9hG4bK-524287-1---88cb65a5723bd495
Call-ID: rDpDbJUXg_dnI87iBuyqFA..
From: <sip:[email protected]>;tag=c6742c10
To: <sip:[email protected]>;tag=z9hG4bK-524287-1---88cb65a5723bd495
CSeq: 1 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1464182775/27b31f1de3708a5c54e7c03550832210",opaque="7a706312205b8f9a",algorithm=md5,qop="auth"
Server: Asterisk
Content-Length:  0


[May 25 15:26:15] VERBOSE[12379] res_pjsip_logger.c: <--- Received SIP request (1145 bytes) from UDP:172.28.131.52:32368 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.28.131.52:32368;branch=z9hG4bK-524287-1---1a7c2f82e4b0c592
Max-Forwards: 70
Contact: <sip:[email protected]:32368;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=c6742c10
Call-ID: rDpDbJUXg_dnI87iBuyqFA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="user",realm="asterisk",nonce="1464182775/27b31f1de3708a5c54e7c03550832210",uri="sip:[email protected];transport=UDP",response="edb8311a8b6d57fc9fccd810dbb109f9",cnonce="52b36302c84eeb986a4e1d45261017a4",nc=00000001,qop=auth,algorithm=md5,opaque="7a706312205b8f9a"
Allow-Events: presence, kpml
Content-Length: 167

v=0
o=Z 0 0 IN IP4 172.28.131.52
s=Z
c=IN IP4 172.28.131.52
t=0 0
m=audio 8000 RTP/AVP 3 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv


[May 25 15:26:15] VERBOSE[12401] res_pjsip_logger.c: <--- Transmitting SIP response (312 bytes) to UDP:172.28.131.52:32368 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.28.131.52:32368;rport=32368;received=172.28.131.52;branch=z9hG4bK-524287-1---1a7c2f82e4b0c592
Call-ID: rDpDbJUXg_dnI87iBuyqFA..
From: <sip:[email protected]>;tag=c6742c10
To: <sip:[email protected]>
CSeq: 2 INVITE
Server: Asterisk
Content-Length:  0


[May 25 15:26:15] VERBOSE[12400] res_pjsip_logger.c: <--- Transmitting SIP request (1029 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 84.130.999.999:5060;rport;branch=z9hG4bKPj992228e9-845f-4806-9b62-8c0a2aa7492a
From: <sip:[email protected]>;tag=eaa7607f-8db6-4ab7-9416-669935d762bb
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 899ac481-d2dd-4e0d-bd5f-3533e81f90d4
CSeq: 18008 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   356

v=0
o=- 1157502723 1157502723 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 my-home.ddns-adresse
t=0 0
m=audio 30012 RTP/AVP 8 111 9 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 25 15:26:15] VERBOSE[12379] res_pjsip_logger.c: <--- Received SIP response (431 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 488 SDP Parameter Error In SIP Request
Via: SIP/2.0/UDP 84.130.999.999:5060;received=84.130.999.999;rport=49419;branch=z9hG4bKPj992228e9-845f-4806-9b62-8c0a2aa7492a
To: <sip:[email protected]>;tag=h7g4Esbg_y8jz97uch317ofzim4wil8i86qmzx7co
From: <sip:[email protected]>;tag=eaa7607f-8db6-4ab7-9416-669935d762bb
Call-ID: 899ac481-d2dd-4e0d-bd5f-3533e81f90d4
CSeq: 18008 INVITE
Content-Length: 0


[May 25 15:26:15] VERBOSE[12400] res_pjsip_logger.c: <--- Transmitting SIP request (437 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 84.130.999.999:5060;rport;branch=z9hG4bKPj992228e9-845f-4806-9b62-8c0a2aa7492a
From: <sip:[email protected]>;tag=eaa7607f-8db6-4ab7-9416-669935d762bb
To: <sip:[email protected]>;tag=h7g4Esbg_y8jz97uch317ofzim4wil8i86qmzx7co
Call-ID: 899ac481-d2dd-4e0d-bd5f-3533e81f90d4
CSeq: 18008 ACK
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 25 15:26:15] VERBOSE[12400] res_pjsip_logger.c: <--- Transmitting SIP response (390 bytes) to UDP:172.28.131.52:32368 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.28.131.52:32368;rport=32368;received=172.28.131.52;branch=z9hG4bK-524287-1---1a7c2f82e4b0c592
Call-ID: rDpDbJUXg_dnI87iBuyqFA..
From: <sip:[email protected]>;tag=c6742c10
To: <sip:[email protected]>;tag=3cc77b43-5878-402e-9bfc-69a5401463ed
CSeq: 2 INVITE
Server: Asterisk
Reason: Q.850;cause=58
Content-Length:  0


[May 25 15:26:15] VERBOSE[12379] res_pjsip_logger.c: <--- Received SIP request (367 bytes) from UDP:172.28.131.52:32368 --->
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.28.131.52:32368;branch=z9hG4bK-524287-1---1a7c2f82e4b0c592
Max-Forwards: 70
To: <sip:[email protected]>;tag=3cc77b43-5878-402e-9bfc-69a5401463ed
From: <sip:[email protected];transport=UDP>;tag=c6742c10
Call-ID: rDpDbJUXg_dnI87iBuyqFA..
CSeq: 2 ACK
Content-Length: 0

rentier-s · 27 Mai 2016

Der SDP Teil passt immer noch nicht. Da müsste in beiden Zeilen die externe IP Adresse stehen.

o=- 369962416 369962416 IN IP4 172.28.131.216
c=IN IP4 my-home.ddns-adresse

Trag bei external_media_address und external_signaling_address manuell Deine aktuelle WAN Adresse ein, mach ein module reload und probier noch mal. Wenn das klappt können wir weiter sehen, wie sich das automatisieren lässt.

jeschero · 30 Mai 2016

Ich habe jetzt meine IP-Adresse eingetragen. Ein- und Ausgehene Anrufe kommen zustande, es wird aber kein Ton übertragen.

xxxx = angerufen/anrufende externe Nummer

yyyy = meine Nummer

zzzz = externe IP

Code:

[May 30 09:35:19] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (838 bytes) from UDP:172.28.131.52:52652 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.28.131.52:52652;branch=z9hG4bK-524287-1---f7715c35dd5a53c5
Max-Forwards: 70
Contact: <sip:[email protected]:52652;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=08224700
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 167

v=0
o=Z 0 0 IN IP4 172.28.131.52
s=Z
c=IN IP4 172.28.131.52
t=0 0
m=audio 8000 RTP/AVP 3 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

[May 30 09:35:19] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (505 bytes) to UDP:172.28.131.52:52652 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.28.131.52:52652;rport=52652;received=172.28.131.52;branch=z9hG4bK-524287-1---f7715c35dd5a53c5
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
From: <sip:[email protected]>;tag=08224700
To: <sip:[email protected]>;tag=z9hG4bK-524287-1---f7715c35dd5a53c5
CSeq: 1 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1464593719/e7b26503175706694942471565c6080f",opaque="25eb47684d376120",algorithm=md5,qop="auth"
Server: Asterisk
Content-Length:  0


[May 30 09:35:19] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (1145 bytes) from UDP:172.28.131.52:52652 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.28.131.52:52652;branch=z9hG4bK-524287-1---516c391612e9ae6e
Max-Forwards: 70
Contact: <sip:[email protected]:52652;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=08224700
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Authorization: Digest username="user",realm="asterisk",nonce="1464593719/e7b26503175706694942471565c6080f",uri="sip:[email protected];transport=UDP",response="e2794b5867274d24a62b52aed216229f",cnonce="bee7f15dc3b205a028399685667959d9",nc=00000001,qop=auth,algorithm=md5,opaque="25eb47684d376120"
Allow-Events: presence, kpml
Content-Length: 167

v=0
o=Z 0 0 IN IP4 172.28.131.52
s=Z
c=IN IP4 172.28.131.52
t=0 0
m=audio 8000 RTP/AVP 3 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv


[May 30 09:35:19] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (312 bytes) to UDP:172.28.131.52:52652 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.28.131.52:52652;rport=52652;received=172.28.131.52;branch=z9hG4bK-524287-1---516c391612e9ae6e
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
From: <sip:[email protected]>;tag=08224700
To: <sip:[email protected]>
CSeq: 2 INVITE
Server: Asterisk
Content-Length:  0


[May 30 09:35:19] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (1014 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP zzzz:5060;rport;branch=z9hG4bKPj245fcd5d-de0e-4c9d-8871-b1980a783351
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
To: <sip:[email protected]>
Contact: <sip:0yyyy@zzzz:5060>
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30825 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   341

v=0
o=- 710577831 710577831 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 zzzz
t=0 0
m=audio 30812 RTP/AVP 9 0 111 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:35:19] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (564 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 407 Proxy Authentication Required 02035034C
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPj245fcd5d-de0e-4c9d-8871-b1980a783351
To: <sip:[email protected]>;tag=h7g4Esbg_85cabe1f0842134a90c20cff8f99c3df
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30825 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce="4B3DDF2242ED4B5700000000F01C4718",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true


[May 30 09:35:19] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (437 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP zzzz:5060;rport;branch=z9hG4bKPj245fcd5d-de0e-4c9d-8871-b1980a783351
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
To: <sip:[email protected]>;tag=h7g4Esbg_85cabe1f0842134a90c20cff8f99c3df
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30825 ACK
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:35:19] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (1303 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP zzzz:5060;rport;branch=z9hG4bKPja6d613e4-58d2-4f0c-a259-00825649c039
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
To: <sip:[email protected]>
Contact: <sip:0yyyy@zzzz:5060>
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30826 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Proxy-Authorization: Digest username="0yyyy", realm="tel.t-online.de", nonce="4B3DDF2242ED4B5700000000F01C4718", uri="sip:[email protected]", response="e7dc7a3c83d941b38e4f32d01da521ef", algorithm=MD5, cnonce="f74caed9-fdf4-48e0-b0cd-3ab5bf84f0a3", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   341

v=0
o=- 710577831 710577831 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 zzzz
t=0 0
m=audio 30812 RTP/AVP 9 0 111 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:35:19] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (356 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPja6d613e4-58d2-4f0c-a259-00825649c039
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30826 INVITE
Content-Length: 0


[May 30 09:35:22] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (946 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPja6d613e4-58d2-4f0c-a259-00825649c039
To: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30826 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Require: 100rel
RSeq: 2
Supported: timer
Content-Type: application/sdp
Content-Length: 232
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE

v=0
o=- 390899381 1284102266 IN IP4 217.0.23.100
s=SS IMS
c=IN IP4 217.0.5.71
t=0 0
m=audio 57156 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=ptime:20
a=sendrecv

[May 30 09:35:22] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (537 bytes) to UDP:217.0.23.100:5060 --->
PRACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP zzzz:5060;rport;branch=z9hG4bKPj832a38df-06e6-4f4b-ae08-d68f636132be
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
To: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30827 PRACK
Route: <sip:217.0.23.100;transport=udp;lr>
RAck: 2 30826 INVITE
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:35:22] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (814 bytes) to UDP:172.28.131.52:52652 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.28.131.52:52652;rport=52652;received=172.28.131.52;branch=z9hG4bK-524287-1---516c391612e9ae6e
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
From: <sip:[email protected]>;tag=08224700
To: <sip:[email protected]>;tag=41e922c7-70a2-41b0-96b5-e62f3f55cbe0
CSeq: 2 INVITE
Server: Asterisk
Contact: <sip:172.28.131.216:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Content-Type: application/sdp
Content-Length:   270

v=0
o=- 0 2 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 172.28.131.216
t=0 0
m=audio 30164 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:35:22] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (628 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPj832a38df-06e6-4f4b-ae08-d68f636132be
To: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30827 PRACK
Supported: timer
Supported: 100rel
Supported: histinfo
Supported: precondition
Supported: norefersub
Content-Length: 0
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE


[May 30 09:35:22] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (621 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPja6d613e4-58d2-4f0c-a259-00825649c039
To: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30826 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Supported: timer
Content-Length: 0
Allow: UPDATE, REFER, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER


[May 30 09:35:22] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (805 bytes) to UDP:172.28.131.52:52652 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.28.131.52:52652;rport=52652;received=172.28.131.52;branch=z9hG4bK-524287-1---516c391612e9ae6e
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
From: <sip:[email protected]>;tag=08224700
To: <sip:[email protected]>;tag=41e922c7-70a2-41b0-96b5-e62f3f55cbe0
CSeq: 2 INVITE
Server: Asterisk
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Contact: <sip:172.28.131.216:5060>
Content-Type: application/sdp
Content-Length:   270

v=0
o=- 0 2 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 172.28.131.216
t=0 0
m=audio 30164 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:35:23] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (890 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPja6d613e4-58d2-4f0c-a259-00825649c039
To: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1285217507-1
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30826 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Supported: timer
Content-Type: application/sdp
Content-Length: 236
Allow: UPDATE, REFER, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER

v=0
o=- 533711753 1285217085 IN IP4 217.0.23.100
s=media server session
t=0 0
m=audio 57156 RTP/AVP 9 101
c=IN IP4 217.0.5.71
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32,36
a=sendrecv
a=ptime:20

[May 30 09:35:23] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (814 bytes) to UDP:172.28.131.52:52652 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.28.131.52:52652;rport=52652;received=172.28.131.52;branch=z9hG4bK-524287-1---516c391612e9ae6e
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
From: <sip:[email protected]>;tag=08224700
To: <sip:[email protected]>;tag=41e922c7-70a2-41b0-96b5-e62f3f55cbe0
CSeq: 2 INVITE
Server: Asterisk
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Contact: <sip:172.28.131.216:5060>
Content-Type: application/sdp
Content-Length:   270

v=0
o=- 0 2 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 172.28.131.216
t=0 0
m=audio 30164 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:35:27] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (1270 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPja6d613e4-58d2-4f0c-a259-00825649c039
To: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30826 INVITE
Contact: <sip:[email protected];transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Session-Expires: 1800;refresher=uas
Supported: timer
Supported: 100rel
Supported: histinfo
Supported: precondition
Supported: norefersub
Content-Type: application/sdp
Content-Length: 232
Session-ID: 18c51395138519951634c34415b4f64c
Authentication-Info: qop=auth,rspauth="980dd03c04565d9ee2b46212c43adffe",cnonce="f74caed9-fdf4-48e0-b0cd-3ab5bf84f0a3",nc=00000001
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE

v=0
o=- 390899381 1284102266 IN IP4 217.0.23.100
s=SS IMS
c=IN IP4 217.0.5.71
t=0 0
m=audio 57156 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=ptime:20
a=sendrecv

[May 30 09:35:27] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (511 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP zzzz:5060;rport;branch=z9hG4bKPjf58d92c1-ef8a-474d-be4e-0aba0197ff8b
From: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
To: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30826 ACK
Route: <sip:217.0.23.100;transport=udp;lr>
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:35:27] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (848 bytes) to UDP:172.28.131.52:52652 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.131.52:52652;rport=52652;received=172.28.131.52;branch=z9hG4bK-524287-1---516c391612e9ae6e
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
From: <sip:[email protected]>;tag=08224700
To: <sip:[email protected]>;tag=41e922c7-70a2-41b0-96b5-e62f3f55cbe0
CSeq: 2 INVITE
Server: Asterisk
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Contact: <sip:172.28.131.216:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   270

v=0
o=- 0 2 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 172.28.131.216
t=0 0
m=audio 30164 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:35:27] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (418 bytes) from UDP:172.28.131.52:52652 --->
ACK sip:172.28.131.216:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.131.52:52652;branch=z9hG4bK-524287-1---ec75f80538e56786
Max-Forwards: 70
Contact: <sip:[email protected]:52652;transport=UDP>
To: <sip:[email protected]>;tag=41e922c7-70a2-41b0-96b5-e62f3f55cbe0
From: <sip:[email protected]>;tag=08224700
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
CSeq: 2 ACK
User-Agent: Z 3.9.32144 r32121
Content-Length: 0


[May 30 09:35:37] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:35:38] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:35:39] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:35:40] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (529 bytes) from UDP:217.0.23.100:5060 --->
BYE sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 60
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7incfykjfhjcpedgmjdhqgndivv
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 30827 BYE
Content-Length: 0
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE


[May 30 09:35:40] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (431 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7incfykjfhjcpedgmjdhqgndivv
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
CSeq: 30827 BYE
Server: Asterisk
Content-Length:  0


[May 30 09:35:40] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (417 bytes) to UDP:172.28.131.52:52652 --->
BYE sip:[email protected]:52652;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.28.131.216:5060;rport;branch=z9hG4bKPja9382af5-4053-4a46-a942-156345825f6c
From: <sip:[email protected]>;tag=41e922c7-70a2-41b0-96b5-e62f3f55cbe0
To: <sip:[email protected]>;tag=08224700
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
CSeq: 20289 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:35:40] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (404 bytes) from UDP:172.28.131.52:52652 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.131.216:5060;rport=5060;branch=z9hG4bKPja9382af5-4053-4a46-a942-156345825f6c
Contact: <sip:[email protected]:52652;transport=UDP>
To: <sip:[email protected]>;tag=08224700
From: <sip:[email protected]>;tag=41e922c7-70a2-41b0-96b5-e62f3f55cbe0
Call-ID: ICrUAiJMKHEpzd5ToGu9jA..
CSeq: 20289 BYE
User-Agent: Z 3.9.32144 r32121
Content-Length: 0


[May 30 09:35:41] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:35:45] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:35:49] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:35:53] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:35:57] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:36:01] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:36:05] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:36:08] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (1261 bytes) from UDP:217.0.23.100:5060 --->
INVITE sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 62
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i8rno8ppkez7cypmetlhjkx1ne
To: <sip:[email protected];user=phone>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 1 INVITE
Contact: <sip:[email protected];transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Min-Se: 900
P-Asserted-Identity: <sip:+49xxxx;[email protected];user=phone>
Session-Expires: 1800
Supported: timer
Supported: 100rel
Supported: histinfo
Content-Type: application/sdp
Content-Length: 255
Session-ID: dc30d3341023fb3c9c24a227cc2acc2d
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE

v=0
o=- 184091498 1330839795 IN IP4 217.0.23.100
s=SS IMS
c=IN IP4 217.0.5.71
t=0 0
m=audio 11644 RTP/AVP 9 8 110
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
a=sendrecv
a=maxptime:40
a=ptime:20

[May 30 09:36:08] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (464 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7i8rno8ppkez7cypmetlhjkx1ne
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65556t1464593768m699229c129351540s2
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
To: <sip:[email protected];user=phone>
CSeq: 1 INVITE
Server: Asterisk
Content-Length:  0


[May 30 09:36:08] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (1070 bytes) to UDP:172.28.131.52:52652 --->
INVITE sip:[email protected]:52652;transport=UDP;rinstance=08cee29f15b9993d SIP/2.0
Via: SIP/2.0/UDP 172.28.131.216:5060;rport;branch=z9hG4bKPj73b2d9a2-7f22-4533-8d51-aa36a86a7e2b
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
To: <sip:[email protected];rinstance=08cee29f15b9993d>
Contact: <sip:[email protected]:5060>
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23547 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   341

v=0
o=- 999307095 999307095 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 172.28.131.216
t=0 0
m=audio 30884 RTP/AVP 9 0 111 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:36:08] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (351 bytes) from UDP:172.28.131.52:52652 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.28.131.216:5060;rport=5060;branch=z9hG4bKPj73b2d9a2-7f22-4533-8d51-aa36a86a7e2b
To: <sip:[email protected];rinstance=08cee29f15b9993d>
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23547 INVITE
Content-Length: 0


[May 30 09:36:08] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (438 bytes) from UDP:172.28.131.52:52652 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.28.131.216:5060;rport=5060;branch=z9hG4bKPj73b2d9a2-7f22-4533-8d51-aa36a86a7e2b
Contact: <sip:[email protected]:52652>
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23547 INVITE
User-Agent: Z 3.9.32144 r32121
Content-Length: 0


[May 30 09:36:08] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (653 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7i8rno8ppkez7cypmetlhjkx1ne
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65556t1464593768m699229c129351540s2
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
CSeq: 1 INVITE
Server: Asterisk
Contact: <sip:zzzz:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Content-Length:  0


[May 30 09:36:09] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (434 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:0yyyy@zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i9luppv7jjuezz7r7zofbo8yix
To: <sip:[email protected]>;tag=c1d6cb51-e805-4d4c-8ded-4765d83b3af0
From: <sip:[email protected]>;tag=h7g4Esbg_p65556t1464593719m661903c129339987s1_1281792725-1025831588
Call-ID: cf7e32e0-48cd-4c37-9f2a-c2ebf2a6f676
CSeq: 1 OPTIONS
Content-Length: 0


[May 30 09:36:12] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (903 bytes) from UDP:172.28.131.52:52652 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.131.216:5060;rport=5060;branch=z9hG4bKPj73b2d9a2-7f22-4533-8d51-aa36a86a7e2b
Require: timer
Contact: <sip:[email protected]:52652>
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23547 INVITE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 167

v=0
o=Z 0 2 IN IP4 172.28.131.52
s=Z
c=IN IP4 172.28.131.52
t=0 0
m=audio 8000 RTP/AVP 0 3 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

[May 30 09:36:12] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (419 bytes) to UDP:172.28.131.52:52652 --->
ACK sip:[email protected]:52652 SIP/2.0
Via: SIP/2.0/UDP 172.28.131.216:5060;rport;branch=z9hG4bKPja438e474-ba3a-46e8-add9-b7b837da3f49
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23547 ACK
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:36:12] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (1047 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7i8rno8ppkez7cypmetlhjkx1ne
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65556t1464593768m699229c129351540s2
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
CSeq: 1 INVITE
Server: Asterisk
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Contact: <sip:zzzz:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   264

v=0
o=- 184091498 1330839797 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 zzzz
t=0 0
m=audio 30672 RTP/AVP 9 8 110
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


[May 30 09:36:12] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (1052 bytes) to UDP:172.28.131.52:52652 --->
INVITE sip:[email protected]:52652 SIP/2.0
Via: SIP/2.0/UDP 172.28.131.216:5060;rport;branch=z9hG4bKPj16e289db-91d9-4df6-92bb-f7439f124201
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
Contact: <sip:[email protected]:5060>
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23548 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   337

v=0
o=- 999307095 999307096 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 217.0.5.71
t=0 0
m=audio 11644 RTP/AVP 9 0 111 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


[May 30 09:36:12] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (903 bytes) from UDP:172.28.131.52:52652 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.131.216:5060;rport=5060;branch=z9hG4bKPj16e289db-91d9-4df6-92bb-f7439f124201
Require: timer
Contact: <sip:[email protected]:52652>
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23548 INVITE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 167

v=0
o=Z 0 3 IN IP4 172.28.131.52
s=Z
c=IN IP4 172.28.131.52
t=0 0
m=audio 8000 RTP/AVP 0 3 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

[May 30 09:36:12] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (419 bytes) to UDP:172.28.131.52:52652 --->
ACK sip:[email protected]:52652 SIP/2.0
Via: SIP/2.0/UDP 172.28.131.216:5060;rport;branch=z9hG4bKPj131e6d1f-6425-4eff-8606-7d4907e2e653
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23548 ACK
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:36:13] VERBOSE[20355] res_pjsip_logger.c: <--- Transmitting SIP response (1047 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7i8rno8ppkez7cypmetlhjkx1ne
Record-Route: <sip:217.0.23.100;transport=udp;lr>
Call-ID: p65556t1464593768m699229c129351540s2
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
CSeq: 1 INVITE
Server: Asterisk
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Contact: <sip:zzzz:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   264

v=0
o=- 184091498 1330839797 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 zzzz
t=0 0
m=audio 30672 RTP/AVP 9 8 110
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:36:13] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (544 bytes) from UDP:217.0.23.100:5060 --->
ACK sip:zzzz:5060 SIP/2.0
Max-Forwards: 64
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ie7z1yo58pc9gmtrkw2wwhuzfx
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 1 ACK
Contact: <sip:[email protected];transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Content-Length: 0


[May 30 09:36:13] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (1152 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP zzzz:5060;rport;branch=z9hG4bKPj8e842ab7-c291-450f-a3c1-cd318176b1b1
From: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Contact: <sip:zzzz:5060>
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 844 INVITE
Route: <sip:217.0.23.100;transport=udp;lr>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
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:   341

v=0
o=- 184091498 1330839798 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 zzzz
t=0 0
m=audio 8000 RTP/AVP 9 0 111 8 3 110
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:36:13] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (606 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 407 Proxy Authentication Required 02035034C
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPj8e842ab7-c291-450f-a3c1-cd318176b1b1
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
From: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 844 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce="D431D90878ED4B57000000009D21A870",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true


[May 30 09:36:13] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (527 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP zzzz:5060;rport;branch=z9hG4bKPj8e842ab7-c291-450f-a3c1-cd318176b1b1
From: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 844 ACK
Route: <sip:217.0.23.100;transport=udp;lr>
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:36:13] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (1445 bytes) to UDP:217.0.23.100:5060 --->
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP zzzz:5060;rport;branch=z9hG4bKPjaf9ccaea-793f-4b64-8cc4-090995e53a0d
From: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Contact: <sip:zzzz:5060>
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 845 INVITE
Route: <sip:217.0.23.100;transport=udp;lr>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Proxy-Authorization: Digest username="0yyyy", realm="tel.t-online.de", nonce="D431D90878ED4B57000000009D21A870", uri="sip:[email protected];transport=udp", response="1614f02c370c6f4b70f5cd93f07d8c84", algorithm=MD5, cnonce="6c03314c-fae2-44ac-84a9-df0514814d4f", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   341

v=0
o=- 184091498 1330839798 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 zzzz
t=0 0
m=audio 8000 RTP/AVP 9 0 111 8 3 110
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:36:14] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (444 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPjaf9ccaea-793f-4b64-8cc4-090995e53a0d
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
From: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 845 INVITE
Content-Length: 0


[May 30 09:36:15] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (1101 bytes) from UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP zzzz:5060;received=zzzz;rport=5060;branch=z9hG4bKPjaf9ccaea-793f-4b64-8cc4-090995e53a0d
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
From: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 845 INVITE
Contact: <sip:[email protected];transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Session-Expires: 1800;refresher=uas
Supported: timer
Content-Type: application/sdp
Content-Length: 232
Authentication-Info: qop=auth,rspauth="0a0a9eec02f2d503e0bc88d1fd2a31e1",cnonce="6c03314c-fae2-44ac-84a9-df0514814d4f",nc=00000001
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE

v=0
o=- 184091498 1330839796 IN IP4 217.0.23.100
s=SS IMS
c=IN IP4 217.0.5.71
t=0 0
m=audio 11644 RTP/AVP 9 110
a=rtpmap:9 G722/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=maxptime:150
a=ptime:20
a=sendrecv

[May 30 09:36:15] WARNING[20356] channel.c: Unable to find a codec translation path: (g722) -> (alaw)
[May 30 09:36:15] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (527 bytes) to UDP:217.0.23.100:5060 --->
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP zzzz:5060;rport;branch=z9hG4bKPj23998b94-9f14-40df-b495-81284bc0f865
From: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 845 ACK
Route: <sip:217.0.23.100;transport=udp;lr>
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:36:23] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7idqgiupqq5o5suh4rhu4r1op47
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 846 OPTIONS
Content-Length: 0


[May 30 09:36:24] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7idqgiupqq5o5suh4rhu4r1op47
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 846 OPTIONS
Content-Length: 0


[May 30 09:36:25] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7idqgiupqq5o5suh4rhu4r1op47
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 846 OPTIONS
Content-Length: 0


[May 30 09:36:27] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7idqgiupqq5o5suh4rhu4r1op47
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 846 OPTIONS
Content-Length: 0


[May 30 09:36:31] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7idqgiupqq5o5suh4rhu4r1op47
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 846 OPTIONS
Content-Length: 0


[May 30 09:36:35] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7idqgiupqq5o5suh4rhu4r1op47
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 846 OPTIONS
Content-Length: 0


[May 30 09:36:39] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7idqgiupqq5o5suh4rhu4r1op47
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 846 OPTIONS
Content-Length: 0


[May 30 09:36:39] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (535 bytes) from UDP:217.0.23.100:5060 --->
BYE sip:zzzz:5060 SIP/2.0
Max-Forwards: 64
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7iesj1zk8tqh3p6jko3t2fvntv7
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 847 BYE
Content-Length: 0
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE


[May 30 09:36:39] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP response (449 bytes) to UDP:217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.23.100:5060;rport=5060;received=217.0.23.100;branch=z9hG4bKg3Zqkv7iesj1zk8tqh3p6jko3t2fvntv7
Call-ID: p65556t1464593768m699229c129351540s2
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
CSeq: 847 BYE
Server: Asterisk
Content-Length:  0


[May 30 09:36:39] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (1056 bytes) to UDP:172.28.131.52:52652 --->
INVITE sip:[email protected]:52652 SIP/2.0
Via: SIP/2.0/UDP 172.28.131.216:5060;rport;branch=z9hG4bKPjbbc52340-31d3-4953-b661-1e23b03194fa
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
Contact: <sip:[email protected]:5060>
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23549 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk
Content-Type: application/sdp
Content-Length:   341

v=0
o=- 999307095 999307097 IN IP4 172.28.131.216
s=Asterisk
c=IN IP4 172.28.131.216
t=0 0
m=audio 30884 RTP/AVP 9 0 111 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[May 30 09:36:39] WARNING[20381][C-00000001] channel.c: Unable to find a codec translation path: (alaw) -> (g722)
[May 30 09:36:39] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (903 bytes) from UDP:172.28.131.52:52652 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.131.216:5060;rport=5060;branch=z9hG4bKPjbbc52340-31d3-4953-b661-1e23b03194fa
Require: timer
Contact: <sip:[email protected]:52652>
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23549 INVITE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.9.32144 r32121
Allow-Events: presence, kpml
Content-Length: 167

v=0
o=Z 0 4 IN IP4 172.28.131.52
s=Z
c=IN IP4 172.28.131.52
t=0 0
m=audio 8000 RTP/AVP 0 3 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

[May 30 09:36:39] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (419 bytes) to UDP:172.28.131.52:52652 --->
ACK sip:[email protected]:52652 SIP/2.0
Via: SIP/2.0/UDP 172.28.131.216:5060;rport;branch=z9hG4bKPj36143df7-2f33-45e0-a7a1-3c935c2c41fa
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23549 ACK
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:36:39] VERBOSE[20356] res_pjsip_logger.c: <--- Transmitting SIP request (419 bytes) to UDP:172.28.131.52:52652 --->
BYE sip:[email protected]:52652 SIP/2.0
Via: SIP/2.0/UDP 172.28.131.216:5060;rport;branch=z9hG4bKPj71071178-089a-49b0-9f30-345db6b567dd
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23550 BYE
Max-Forwards: 70
User-Agent: Asterisk
Content-Length:  0


[May 30 09:36:39] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP response (430 bytes) from UDP:172.28.131.52:52652 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.131.216:5060;rport=5060;branch=z9hG4bKPj71071178-089a-49b0-9f30-345db6b567dd
Contact: <sip:[email protected]:52652>
To: <sip:[email protected];rinstance=08cee29f15b9993d>;tag=df1ca730
From: <sip:[email protected]>;tag=03a7ab87-d9cb-4a27-be22-f3d085078feb
Call-ID: ac095721-244d-4fab-a9f2-439ab5192abb
CSeq: 23550 BYE
User-Agent: Z 3.9.32144 r32121
Content-Length: 0


[May 30 09:36:43] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7idqgiupqq5o5suh4rhu4r1op47
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 846 OPTIONS
Content-Length: 0


[May 30 09:36:47] VERBOSE[20355] res_pjsip_logger.c: <--- Received SIP request (444 bytes) from UDP:217.0.23.100:5060 --->
OPTIONS sip:zzzz:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7idqgiupqq5o5suh4rhu4r1op47
To: <sip:[email protected];user=phone>;tag=2eb6c024-f75b-4180-8fa7-a1b269cdc954
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65556t1464593768m699229c129351540s1_1330840567-229013524
Call-ID: p65556t1464593768m699229c129351540s2
CSeq: 846 OPTIONS
Content-Length: 0

rentier-s · 31 Mai 2016

Scheinbar mag PJSIP die WAN Adresse nicht als Origin. Das einzige was mir sonst noch auffällt ist, dass der Client zwar Codecs angibt, aber keine Attribute dazu, weiß aber ehrlich gesagt nicht ob das ein Problem ist.

[Problem] Fehlermeldung Anmeldung: No registered publish handler for event presence

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