marco Posted July 19, 2012 Report Share Posted July 19, 2012 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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted July 19, 2012 Report Share Posted July 19, 2012 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. Quote Link to comment Share on other sites More sharing options...
katerina Posted July 26, 2012 Report Share Posted July 26, 2012 Versuchen Sie bitte in den Trunkeinstellungen das "CLIP Standard/Anzeige von Nummern:" auf "Keine Anzeige" zu setzten. Wenn das auch nicht geht bitte das snom ONE Log nochmals machen wie hier beschrieben: http://wiki.snomone.com/index.php?title=Snom_ONE_log Quote Link to comment Share on other sites More sharing options...
gifti Posted September 11, 2017 Report Share Posted September 11, 2017 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 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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 14, 2017 Report Share Posted September 14, 2017 Wir müssen das einfach ins Drop-Down übernehmen. Dann sollte es sehr einfach sein, die ISDN-SIP-Umstellung hinzubekommen :-) Quote Link to comment Share on other sites More sharing options...
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