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Ausgehende Rufe nicht möglich

marco

By marco
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marco Newbie

marco

Posted
Posted

Hallo Forum,

 

wir wollen gerad eine Snom One mit unseren Deutschland LAN Rufnummern verquicken. Installiert ist ssie auf einer VM Server 2008 R2 64 Bit.

Eingehende Anrufe funktionieren prima. Wenn wir nun von dem intern registrierten Snom 870 einen Anruf nach Draußen tätigen wollen bekommen wir folgenden Fehler im Log der One angezeigt:

 

 

 

[5] 2012/07/19 19:17:56: Dialplan "dlan": Match 045xxxxxx@192.168.178.215 to sip:045xxxxxx3@9ol10001.tel.deutschland-lan.de;user=phone on trunk 043xx-59xxx-xx

[5] 2012/07/19 19:17:56: Charge user 40 for redirecting calls

[5] 2012/07/19 19:17:56: set codec: codec pcma/8000 is set to call-leg 6

[5] 2012/07/19 19:17:56: INVITE Response 400 Bad header field: p-asserted-identity: Terminate 4a9e926f@pbx

 

 

und auf dem 870:

 

Sent to udp:192.168.178.215:5060 at 19/7/2012 19:20:00:932 (1047 bytes):

 

INVITE sip:045xxxxxxx@192.168.178.215;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-ypimhkun4zd3;rport

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.178.87:3072;line=p7s0pz2y>;reg-id=1

X-Serialnumber: 000413414E70

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom870/8.4.31

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Content-Type: application/sdp

Content-Length: 248

 

v=0

o=root 2094392128 2094392128 IN IP4 192.168.178.87

s=call

c=IN IP4 192.168.178.87

t=0 0

m=audio 64080 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

Received from udp:192.168.178.215:5060 at 19/7/2012 19:20:00:938 (524 bytes):

 

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-ypimhkun4zd3;rport=3072

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 1 INVITE

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

WWW-Authenticate: Digest realm="192.168.178.215",nonce="f0d046e717e0496b85a20c21a1dde6f4",domain="sip:045xxxxxxx@192.168.178.215;user=phone",algorithm=MD5

Content-Length: 0

 

Sent to udp:192.168.178.215:5060 at 19/7/2012 19:20:00:941 (409 bytes):

 

ACK sip:045xxxxxxx@192.168.178.215;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-ypimhkun4zd3;rport

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.178.87:3072;line=p7s0pz2y>;reg-id=1

Content-Length: 0

 

Sent to udp:192.168.178.215:5060 at 19/7/2012 19:20:00:948 (1257 bytes):

 

INVITE sip:045xxxxxxx@192.168.178.215;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-76rk11l1m892;rport

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 2 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.178.87:3072;line=p7s0pz2y>;reg-id=1

X-Serialnumber: 000413414E70

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom870/8.4.31

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Authorization: Digest username="40",realm="192.168.178.215",nonce="f0d046e717e0496b85a20c21a1dde6f4",uri="sip:045xxxxxxx@192.168.178.215;user=phone",response="0ef875e358b74ebb3bc68abf19ff635f",algorithm=MD5

Content-Type: application/sdp

Content-Length: 248

 

v=0

o=root 2094392128 2094392128 IN IP4 192.168.178.87

s=call

c=IN IP4 192.168.178.87

t=0 0

m=audio 64080 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

Received from udp:192.168.178.215:5060 at 19/7/2012 19:20:00:951 (300 bytes):

 

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-76rk11l1m892;rport=3072

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 2 INVITE

Content-Length: 0

 

Received from udp:192.168.178.215:5060 at 19/7/2012 19:20:01:344 (877 bytes):

 

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-76rk11l1m892;rport=3072

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 2 INVITE

Contact: <sip:40@192.168.178.215:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 271

 

v=0

o=- 12580 12580 IN IP4 192.168.178.215

s=-

c=IN IP4 192.168.178.215

t=0 0

m=audio 51782 RTP/AVP 8 0 101

a=rtpmap:8 pcma/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

Sent to udp:192.168.178.215:5060 at 19/7/2012 19:20:01:492 (557 bytes):

 

PRACK sip:40@192.168.178.215:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-si31wjz6m5n6;rport

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 3 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.178.87:3072;line=p7s0pz2y>;reg-id=1

RAck: 1 2 INVITE

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

Received from udp:192.168.178.215:5060 at 19/7/2012 19:20:01:881 (568 bytes):

 

SIP/2.0 400 Bad header field: p-asserted-identity

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-76rk11l1m892;rport=3072

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 2 INVITE

Contact: <sip:40@192.168.178.215:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Content-Length: 0

 

Sent to udp:192.168.178.215:5060 at 19/7/2012 19:20:01:884 (409 bytes):

 

ACK sip:045xxxxxxx@192.168.178.215;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-76rk11l1m892;rport

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.178.87:3072;line=p7s0pz2y>;reg-id=1

Content-Length: 0

 

Sent to udp:192.168.178.215:5060 at 19/7/2012 19:20:02:332 (557 bytes):

 

PRACK sip:40@192.168.178.215:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-si31wjz6m5n6;rport

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 3 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.178.87:3072;line=p7s0pz2y>;reg-id=1

RAck: 1 2 INVITE

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

Received from udp:192.168.178.215:5060 at 19/7/2012 19:20:02:336 (568 bytes):

 

SIP/2.0 400 Bad header field: p-asserted-identity

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-76rk11l1m892;rport=3072

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 2 INVITE

Contact: <sip:40@192.168.178.215:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Content-Length: 0

 

Sent to udp:192.168.178.215:5060 at 19/7/2012 19:20:02:337 (409 bytes):

 

ACK sip:045xxxxxxx@192.168.178.215;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-76rk11l1m892;rport

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.178.87:3072;line=p7s0pz2y>;reg-id=1

Content-Length: 0

 

Received from udp:192.168.178.215:5060 at 19/7/2012 19:20:03:137 (384 bytes):

 

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-si31wjz6m5n6;rport=3072

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 3 PRACK

Contact: <sip:40@192.168.178.215:5060>

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Content-Length: 0

 

Received from udp:192.168.178.215:5060 at 19/7/2012 19:20:03:154 (384 bytes):

 

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-si31wjz6m5n6;rport=3072

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=lv08fwwt2o

To: <sip:045xxxxxxx@192.168.178.215;user=phone>;tag=f9ac2a2e14

Call-ID: 8a7e263cc899-kqej93pdv9hf

CSeq: 3 PRACK

Contact: <sip:40@192.168.178.215:5060>

User-Agent: snomONE/4.5.0.1090 Epsilon Geminids

Content-Length: 0

 

Sent to udp:192.168.178.215:5060 at 19/7/2012 19:20:07:688 (738 bytes):

 

REGISTER sip:192.168.178.215 SIP/2.0

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-6ifug64kfi1c;rport

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=n7fu9qhh8x

To: "043xxxxxxx" <sip:40@192.168.178.215>

Call-ID: 2770263cd189-pp6g6syo7bfn

CSeq: 101 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.178.87:3072;line=p7s0pz2y>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:1159d8e2-c267-46e0-8830-7aba761d5d7d>";audio;mobility="fixed";duplex="full";description="snom870";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"

User-Agent: snom870/8.4.31

Allow-Events: dialog

X-Real-IP: 192.168.178.87

Supported: path, gruu

Expires: 3600

Content-Length: 0

 

Received from udp:192.168.178.215:5060 at 19/7/2012 19:20:07:692 (423 bytes):

 

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.178.87:3072;branch=z9hG4bK-6ifug64kfi1c;rport=3072

From: "043xxxxxxx" <sip:40@192.168.178.215>;tag=n7fu9qhh8x

To: "043xxxxxxx" <sip:40@192.168.178.215>;tag=5775b680aa

Call-ID: 2770263cd189-pp6g6syo7bfn

CSeq: 101 REGISTER

Contact: <sip:40@192.168.178.87:3072;line=p7s0pz2y>;expires=360

Supported: path

Server: snomONE/4.5.0.1090 Epsilon Geminids

Content-Length: 0

 

Hoffe uns kann einer von euch weiterhelfen.

 

Viele Grüße

Marco

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Vodia PBX Grand Master

Vodia PBX

Posted
Posted

Ich würde mal zu 99 % vermuten dass das Problem auf der Trunk (Leitung)-Seite liegt. Die PBX generiert selbst nicht solche Fehlermeldungen, sondern reicht sie nur durch vom Service-Provider. Bei 4.5 kann man recht einfach die Header im Trunk ändern; vermutlich reicht es aus, P-Asserted-Identity einfach leer zu machen. Welcher Service Provider ist im Einsatz?

 

Am Rande auch hier nochmal der Hinweis: Leute, bitte verwendet doch Plug and Play für die snom Telefone. Damit wird alles so eingestellt dass es keinen Ärger gibt. Hier wird UDP für SIP zwischen PBX und Telefon verwendet, was vor allem bei längeren Nachrichten problematisch ist und darüberhinaus auch noch unsicher.

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  • 5 years later...
gifti Newbie

gifti

Posted
Posted

Hallo Marco, hallo Vodia-Support-Team,

darf ich fragen, mit welchen Einstellungen du den der DeutschlandLAN SIP-Trunk zum Laufen gebracht hast?
Ich habe das Problem, dass ich keine Antwort (Error 408) vom Registar der Telekom erhalte.
  

8] 2017/09/11 12:17:35: Trunk 15: Preparing for re-registration
[8] 2017/09/11 12:17:35: Trunk 15: sending discover message for sip-trunk.telekom.de
[8] 2017/09/11 12:17:35: Trunk 15 (Telekom SIP-Trunk) is associated with the following addresses: tls:217.0.26.35:5061 tls:217.0.26.37:5061 tls:217.0.26.69:5061
[8] 2017/09/11 12:17:35: Trunk Telekom SIP-Trunk: Sending registration to reg.sip-trunk.telekom.de
[8] 2017/09/11 12:17:36: Packet authenticated by transport layer
[8] 2017/09/11 12:18:07: Last message repeated 48 times
[5] 2017/09/11 12:18:07: Registration on trunk 15 (Telekom SIP-Trunk) failed with code 408. Retry in 60 seconds

Die Firewall ist offen. Auch die DNS Auflösung über NAPTR -> SRV -> TypA auf die drei Telekom-IPs scheint zu klappen.
Ich habe einen Trunk vom Typ "SIP-Registrierung" angelegt und folgende Felder befüllt:

(Deutsche Telekom / Vodia)
Outbound-Proxy ==> Proxy Adresse
Registar ==> Domäne
Registrierungsrufnummer ==> Angezeigter Name
Telefonie Benutzername ==> Konto
Telefonie Benutzername  ==> Benutzername

dlan.jpg

Vodia Version: 5.1.1 (CentOS64)

Die Telekom meint, es könnte eventuell daran liegen, dass die TLS Verschlüsselung nicht aufgebaut werden kann!? Muss ich mein System Updaten? Ich nutze auf unsere Systemwebseite noch das SSI {ssi call_history domain}. Damit kann jeder Nutzer schnell alle Gespräche aller Nutzer auf dem System einsehen. Bei neueren Vodia Versionen wurde dieses Feature (ich nehme an aus Sicherheitsgründen) leider gestrichen ...

Vielen Dank und Beste Grüße
Gifti / Wolfgang

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