Tom Tom

Nachfolgend habe ich die Konfig (aktuell und ohne Funktion/Registrierung) des Cisco ATA gepostet, sieht hier jemand einen Fehler? UIPassword:0 UseTftp:1 TftpURL:xx.xx.xx.xx CfgInterval:3600 EncryptKey:0 upgradecode:0,0x301,0x0400,0x0200,0.0.0.0,69,0,none dhcp:1 StaticIp:0 StaticRoute:0 StaticNetMask:0 DNS1IP:xx.xx.xx.xx DNS2IP:0.0.0.0 NTPIP:xx.xx.xx.xx AltNTPIP:0.0.0.0 VLANSetting:0x0000002b PortsSetting:0x00000044 L2KeepAlive:0 GkOrProxy:0 Proxy:xx.xx.xx.xx AltGk:0 SecProxy:0 AltGkTimeOut:0 SecProxyTimeOut:0 UID0:50 UID1:0 PWD0:abcona PWD1:abcona LoginID0:0 LoginID1:0 UseLoginID:0 SIPPort:5060 SIPRegInterval:120 SIPRegOn:1 MaxRedirect:5 SipOutBoundProxy:xx.xx.xx.xx NATIP:0 NatServer:0 NatTimer:0x00000000 MsgRetryLimits:0x00000000 SessionTimer:0x00000000 SessionInterval:1800 MinSessionInterval:1800 DisplayName0:01 DisplayName1:02 ACRDN:0 MediaPort:16384 RxCodec:1 TxCodec:1 LBRCodec:0 AudioMode:0x00150015 NumTxFrames:2 TOS:0x000068B8 PaidFeatures:0xffffffff CallFeatures:0xffffffff CallCmd:Af;AH;BS;NA;CS;NA;Df;EB;Ff;EP;Kf;EFh;HH;Jf;AFh;EQ;I*67;gA*82;fA#90v#;OI;H#72v#;bA#74v#;cA#75v#;dA#73;eA*67;gA*82;fA*70;iA*69;DA*99;xA*77;lA*87;mA;Uh;GQ; FeatureTimer:0x00000000 FeatureTimer2:0x0000001e SigTimer:0x01418564 ConnectMode:0x00060400 OpFlags:0x00000002 TimeZone:17 CallerIdMethod:0x00019e60 Polarity:0 FXSInputLevel:-1 FXSOutputLevel:-4 DialTone:2,31538,30831,1380,1740,1,0,0,1000,0,0 BusyTone:2,30467,28959,1191,1513,0,4000,4000,0,0,0 ReorderTone:2,30467,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0 RingBackTone:2,30831,30467,1943,2111,0,16000,32000,0,0,0 CallWaitTone:1,30831,0,5493,0,0,2400,2400,4800,0,0 AlertTone:1,30467,0,5970,0,0,480,480,1920,0,0 SITone:2,30467,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0 RingOnOffTime:2,4,25 DialPlan:*St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.- IPDialPlan:1 NPrintf:0 TraceFlags:0x00000000 SyslogIP:0.0.0.0.514 SyslogCtrl:0x00000000

Aber dann müsste das doch nur für Externe Anrufe zutreffen!?! ich habe das aber auch beim internen weiterverbinden, das der Anruf zwar durchgereicht wird aber dann ist beim anderen Internteilnehmer stille in der Leitung verbindung ist aber da. Wenn ich Ihn wieder zurückhole kann sogar sein es geht wieder mit Ton.

Hallo zusammen, mal wieder ein seltsames Phänomen... teileweise passiert es wenn ich einen abgehenden Anruf tätige wird zwar eine Verbindung aufgebaut aber weder ich noch der Angerufene hört den Anderen. Die Verbindung kommt aber zustande da der Angerufene mir sagt das ich doch eben schon angerufen hätte aber nichts zu hören war. Dieses Phänomen habe ich zeitweise auch beim internen Vermitteln. Externen Anrufer ander Strippe, Rückfrage intern klappt noch aber beim Transfer zum anderen Internen Teilnehmer ist die Verbindung noch da aber es ist absolute Stille im Hörer.. Hat dazu jemand eine Idee ?? Ich vergaß: SnomOne blue + Snom 320 Telefone, externe Anbindung per direkt SIP an den SBC bei Colt Telecom mit 20 Nutzkanälen gleichzeitig.

Hallo, wir haben das Problem das wir zwar intern telefonieren können, abhaben Nebenstelle wählen und los. Wenn ich aber einen Anruf von extern an einem Apparat entgegennehme und dann mit Hold parke kann ich nicht die interne Nebenstelle anrufen Meldung Service unavailible Wer kennt das Problem? schon mal Danke im Voraus

Telefonieren rein und raus geht.. 1. Fehler bei Colt, falsche IP hinterlegt. 2. Fehler Port im Router nicht freigegeben. Danke für die Hilfe

Mittlerweile hat der Provider nachgebessert jetzt gehen die Anrufe wenigtens raus aber es kommt kein Ruf an.

Hier das Log nach Anleitung [9] 2012/03/30 12:35:08: Last message repeated 2 times [7] 2012/03/30 12:35:08: SIP Rx tls:10.10.0.50:2629: REGISTER sip:pbx.company.com SIP/2.0 Via: SIP/2.0/TLS 10.10.0.50:2629;branch=z9hG4bK-mc1995cjtofo;rport From: "Forty One" <sip:41@pbx.company.com>;tag=8pj4af5nqe To: "Forty One" <sip:41@pbx.company.com> Call-ID: 3c26702249e2-bi3qpnp77vpo CSeq: 233216 REGISTER Max-Forwards: 70 Contact: <sip:41@10.10.0.50:2629;transport=tls;line=xdtdbgrm>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:a9c7202a-7651-4857-bad0-2b032eb6bfc7>" User-Agent: snom320/8.4.18 Allow-Events: dialog X-Real-IP: 10.10.0.50 Supported: path, gruu WWW-Contact: <http://10.10.0.50:80> WWW-Contact: <https://10.10.0.50:443> Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2012/03/30 12:35:08: Packet authenticated by transport layer [7] 2012/03/30 12:35:08: SIP Tx tls:10.10.0.50:2629: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.10.0.50:2629;branch=z9hG4bK-mc1995cjtofo;rport=2629 From: "Forty One" <sip:41@pbx.company.com>;tag=8pj4af5nqe To: "Forty One" <sip:41@pbx.company.com>;tag=cf8d036600 Call-ID: 3c26702249e2-bi3qpnp77vpo CSeq: 233216 REGISTER Contact: <sip:41@10.10.0.50:2629;transport=tls;line=xdtdbgrm>;expires=179 Supported: path Content-Length: 0 [7] 2012/03/30 12:35:09: SIP Rx tls:10.10.0.49:2059: SUBSCRIBE sip:10.10.0.11:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-sbpud3410dc2;rport From: <sip:49@localhost>;tag=6ny6hja4qh To: <sip:49@localhost;user=phone>;tag=226a6820c5 Call-ID: 3c267023801b-gdsys4unv55b CSeq: 34 SUBSCRIBE Max-Forwards: 70 Contact: <sip:49@10.10.0.49:2059;transport=tls;line=yoxynt0d>;reg-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom320/8.4.18 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2012/03/30 12:35:09: Packet authenticated by transport layer [7] 2012/03/30 12:35:09: SIP Tx tls:10.10.0.49:2059: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-sbpud3410dc2;rport=2059 From: <sip:49@localhost>;tag=6ny6hja4qh To: <sip:49@localhost;user=phone>;tag=226a6820c5 Call-ID: 3c267023801b-gdsys4unv55b CSeq: 34 SUBSCRIBE Contact: <sip:10.10.0.11:5061;transport=tls> Expires: 182 Content-Length: 0 [7] 2012/03/30 12:35:10: SIP Rx tls:10.10.0.49:2059: INVITE sip:01721009776@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg To: <sip:01721009776@localhost;user=phone> Call-ID: 3c267bc0976c-ccg452e9pmln CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:49@10.10.0.49:2059;transport=tls;line=yoxynt0d>;reg-id=1 X-Serialnumber: 0004133800E1 P-Key-Flags: keys="3" User-Agent: snom320/8.4.18 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 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 520 v=0 o=root 2107876324 2107876324 IN IP4 10.10.0.49 s=call c=IN IP4 10.10.0.49 t=0 0 m=audio 58144 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZbHe/ofcDrAp8sW5MGIOmEgsfHFJnT1usyc/STE0 a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [8] 2012/03/30 12:35:10: Packet authenticated by transport layer [8] 2012/03/30 12:35:10: Allocating call port 62, SIP call id 3c267bc0976c-ccg452e9pmln [9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:61556 [9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:57720 [9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:62340 [9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:62341 [9] 2012/03/30 12:35:10: UDP: Opening socket on [::]:62340 [9] 2012/03/30 12:35:10: UDP: Opening socket on [::]:62341 [8] 2012/03/30 12:35:10: Could not find a trunk (2 trunks) [9] 2012/03/30 12:35:10: Using outbound proxy sip:10.10.0.49:2059;transport=tls because of flow-label [9] 2012/03/30 12:35:10: Last message repeated 3 times [7] 2012/03/30 12:35:10: SIP Tx tls:10.10.0.49:2059: SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport=2059 From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec Call-ID: 3c267bc0976c-ccg452e9pmln CSeq: 1 INVITE Content-Length: 0 [7] 2012/03/30 12:35:10: Set packet length to 20 [6] 2012/03/30 12:35:10: Sending RTP for 3c267bc0976c-ccg452e9pmln to 10.10.0.49:58144, codec not set yet [8] 2012/03/30 12:35:10: Incoming call: Request URI sip:01721009776@localhost;user=phone, To is <sip:01721009776@localhost;user=phone> [8] 2012/03/30 12:35:10: Call from an user 49 [8] 2012/03/30 12:35:10: To is <sip:01721009776@localhost;user=phone>, user 0, domain 1 [8] 2012/03/30 12:35:10: From user 49 [8] 2012/03/30 12:35:10: Set the To domain based on From user 49@localhost [8] 2012/03/30 12:35:10: Call state for call object 27: idle [9] 2012/03/30 12:35:10: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 01721009776@localhost [5] 2012/03/30 12:35:10: Dialplan "Standard Schema": Match 01721009776@localhost to <sip:01721009776@217.110.34.74;user=phone> on trunk Colt [8] 2012/03/30 12:35:10: Play audio_moh/noise.wav [7] 2012/03/30 12:35:10: set_codecs: for 3c267bc0976c-ccg452e9pmln codecs "", codec_preference count 6 [8] 2012/03/30 12:35:10: Allocating call port 63, SIP call id 9d46514b@pbx [9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:59700 [9] 2012/03/30 12:35:10: UDP: Opening socket on 0.0.0.0:59701 [9] 2012/03/30 12:35:10: UDP: Opening socket on [::]:59700 [9] 2012/03/30 12:35:10: UDP: Opening socket on [::]:59701 [7] 2012/03/30 12:35:10: set_codecs: for 9d46514b@pbx codecs "", codec_preference count 6 [8] 2012/03/30 12:35:10: call port 63: state code from 0 to 100 [9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec pcmu/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec pcma/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec g722/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec g726-32/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: adding codec gsm/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 9d46514b@pbx: codec_preference size 6, available codecs size 6 [9] 2012/03/30 12:35:10: Resolve 45128: url sip:217.110.34.74 [9] 2012/03/30 12:35:10: Resolve 45128: udp 217.110.34.74 5060 [7] 2012/03/30 12:35:10: SIP Tx udp:217.110.34.74:5060: INVITE sip:00491721009776@217.110.34.74;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;rport From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943 To: <sip:00491721009776@217.110.34.74;user=phone> Call-ID: 9d46514b@pbx CSeq: 10383 INVITE Max-Forwards: 70 Contact: <sip:703173588@10.10.0.11:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.3.0.5021 Content-Type: application/sdp Content-Length: 323 v=0 o=- 34232 34232 IN IP4 10.10.0.11 s=- c=IN IP4 10.10.0.11 t=0 0 m=audio 59700 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2012/03/30 12:35:10: SIP Rx tls:10.10.0.27:2591: SUBSCRIBE sip:10.10.0.11:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.10.0.27:2591;branch=z9hG4bK-psxzkrknaqfc;rport From: <sip:42@pbx.company.com>;tag=4njk5y6n67 To: <sip:42@pbx.company.com;user=phone>;tag=20b41650fd Call-ID: 3cb37e180a0b-alvbnft4ukno CSeq: 101800 SUBSCRIBE Max-Forwards: 70 Contact: <sip:42@10.10.0.27:2591;transport=tls;line=zsuhx64g>;reg-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom320/8.4.18 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2012/03/30 12:35:10: Packet authenticated by transport layer [7] 2012/03/30 12:35:10: SIP Tx tls:10.10.0.27:2591: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.10.0.27:2591;branch=z9hG4bK-psxzkrknaqfc;rport=2591 From: <sip:42@pbx.company.com>;tag=4njk5y6n67 To: <sip:42@pbx.company.com;user=phone>;tag=20b41650fd Call-ID: 3cb37e180a0b-alvbnft4ukno CSeq: 101800 SUBSCRIBE Contact: <sip:10.10.0.11:5061;transport=tls> Expires: 179 Content-Length: 0 [8] 2012/03/30 12:35:10: call port 62: state code from 0 to 183 [7] 2012/03/30 12:35:10: Set packet length to 20 [9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec pcmu/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec pcma/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec g722/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec g726-32/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: adding codec gsm/8000 to available list [9] 2012/03/30 12:35:10: update_codecs for 3c267bc0976c-ccg452e9pmln: codec_preference size 6, available codecs size 6 [6] 2012/03/30 12:35:10: Codec pcmu/8000 is chosen for call id 3c267bc0976c-ccg452e9pmln [7] 2012/03/30 12:35:10: SIP Tx tls:10.10.0.49:2059: SIP/2.0 183 Session Progress Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport=2059 From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec Call-ID: 3c267bc0976c-ccg452e9pmln CSeq: 1 INVITE Contact: <sip:49@10.10.0.11:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.3.0.5021 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 417 v=0 o=- 2505 2505 IN IP4 10.10.0.11 s=- c=IN IP4 10.10.0.11 t=0 0 m=audio 62340 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:GrNA0lN6ROxjsRLIZVwmZ7edO8VjfxIdM0ek+obz a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/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 [7] 2012/03/30 12:35:10: SIP Rx udp:217.110.34.74:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;received=213.61.108.147;rport=22980 From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943 To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5 Call-ID: 9d46514b@pbx CSeq: 10383 INVITE Content-Length: 0 [9] 2012/03/30 12:35:10: Message repetition, packet dropped [7] 2012/03/30 12:35:10: SIP Rx tls:10.10.0.49:2059: PRACK sip:49@10.10.0.11:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-ijk623j7njd5;rport From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec Call-ID: 3c267bc0976c-ccg452e9pmln CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:49@10.10.0.49:2059;transport=tls;line=yoxynt0d>;reg-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Proxy-Require: buttons Content-Length: 0 [8] 2012/03/30 12:35:10: Packet authenticated by transport layer [7] 2012/03/30 12:35:10: SIP Tx tls:10.10.0.49:2059: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-ijk623j7njd5;rport=2059 From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec Call-ID: 3c267bc0976c-ccg452e9pmln CSeq: 2 PRACK Contact: <sip:49@10.10.0.11:5061;transport=tls> User-Agent: snom-PBX/2011-4.3.0.5021 Content-Length: 0 [8] 2012/03/30 12:35:10: SRTP MAC mismatch: f9318abd != 4f4d0000 [7] 2012/03/30 12:35:10: Discard SRTCP packet from 10.10.0.49:58145 with wrong MAC [7] 2012/03/30 12:35:11: SIP Rx udp:217.110.34.74:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;received=213.61.108.147;rport=22980 From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943 To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5 Call-ID: 9d46514b@pbx CSeq: 10383 INVITE Contact: <sip:00491721009776@217.110.34.74:5060> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Require: 100rel RSeq: 14218 Content-Length: 235 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 28998 14816 IN IP4 217.110.34.74 s=SIP Media Capabilities c=IN IP4 217.110.34.73 t=0 0 m=audio 25878 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 [7] 2012/03/30 12:35:11: Set packet length to 20 [6] 2012/03/30 12:35:11: Codec pcma/8000 is chosen for call id 9d46514b@pbx [6] 2012/03/30 12:35:11: Sending RTP for 9d46514b@pbx to 217.110.34.73:25878, codec pcma/8000 [9] 2012/03/30 12:35:11: Resolve 45132: url sip:00491721009776@217.110.34.74:5060 [9] 2012/03/30 12:35:11: Resolve 45132: udp 217.110.34.74 5060 [7] 2012/03/30 12:35:11: SIP Tx udp:217.110.34.74:5060: PRACK sip:00491721009776@217.110.34.74:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-2c16c59aa7f2b83e02538a5092ae3c3c;rport From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943 To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5 Call-ID: 9d46514b@pbx CSeq: 10384 PRACK Max-Forwards: 70 Contact: <sip:703173588@10.10.0.11:5060;transport=udp> RAck: 14218 10383 INVITE Content-Length: 0 [8] 2012/03/30 12:35:11: Call state for call object 27: alerting [8] 2012/03/30 12:35:11: call port 62: state code from 183 to 183 [8] 2012/03/30 12:35:11: Last message repeated 2 times [7] 2012/03/30 12:35:11: 3c267bc0976c-ccg452e9pmln: RTP pass-through mode [7] 2012/03/30 12:35:11: 9d46514b@pbx: RTP pass-through mode [6] 2012/03/30 12:35:11: Different Codecs (local pcmu/8000, remote pcma/8000), callid 3c267bc0976c-ccg452e9pmln, falling back to transcoding [7] 2012/03/30 12:35:11: SIP Rx udp:217.110.34.74:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-2c16c59aa7f2b83e02538a5092ae3c3c;received=213.61.108.147;rport=22980 From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943 To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5 Call-ID: 9d46514b@pbx CSeq: 10384 PRACK Content-Length: 0 [7] 2012/03/30 12:35:11: Call 9d46514b@pbx: Clear last request [7] 2012/03/30 12:35:12: SIP Rx tls:10.10.0.49:2059: CANCEL sip:01721009776@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg To: <sip:01721009776@localhost;user=phone> Call-ID: 3c267bc0976c-ccg452e9pmln CSeq: 1 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Proxy-Require: buttons Content-Length: 0 [8] 2012/03/30 12:35:12: Packet authenticated by transport layer [7] 2012/03/30 12:35:12: SIP Tx tls:10.10.0.49:2059: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport=2059 From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec Call-ID: 3c267bc0976c-ccg452e9pmln CSeq: 1 CANCEL Contact: <sip:49@10.10.0.11:5061;transport=tls> User-Agent: snom-PBX/2011-4.3.0.5021 Content-Length: 0 [7] 2012/03/30 12:35:12: SIP Tx tls:10.10.0.49:2059: SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport=2059 From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec Call-ID: 3c267bc0976c-ccg452e9pmln CSeq: 1 INVITE Contact: <sip:49@10.10.0.11:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.3.0.5021 Content-Length: 0 [8] 2012/03/30 12:35:12: Remove leg 50: call port 62, SIP call id 3c267bc0976c-ccg452e9pmln [8] 2012/03/30 12:35:12: call port 63: state code from 100 to 486 [9] 2012/03/30 12:35:12: Resolve 45135: udp 217.110.34.74 5060 [7] 2012/03/30 12:35:12: SIP Tx udp:217.110.34.74:5060: CANCEL sip:00491721009776@217.110.34.74;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;rport From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943 To: <sip:00491721009776@217.110.34.74;user=phone> Call-ID: 9d46514b@pbx CSeq: 10383 CANCEL Max-Forwards: 70 Content-Length: 0 [7] 2012/03/30 12:35:12: SIP Rx udp:217.110.34.74:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;received=213.61.108.147;rport=22980 From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943 To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5 Call-ID: 9d46514b@pbx CSeq: 10383 CANCEL Content-Length: 0 [7] 2012/03/30 12:35:12: Call 9d46514b@pbx: Clear last request [7] 2012/03/30 12:35:12: SIP Rx udp:217.110.34.74:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;received=213.61.108.147;rport=22980 From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943 To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5 Call-ID: 9d46514b@pbx CSeq: 10383 INVITE Content-Length: 0 [7] 2012/03/30 12:35:12: Call 9d46514b@pbx: Clear last INVITE [9] 2012/03/30 12:35:12: Resolve 45136: url sip:00491721009776@217.110.34.74:5060 [9] 2012/03/30 12:35:12: Resolve 45136: udp 217.110.34.74 5060 [7] 2012/03/30 12:35:12: SIP Tx udp:217.110.34.74:5060: ACK sip:00491721009776@217.110.34.74:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.11:5060;branch=z9hG4bK-99e91d815ffefe3d479eb242453a952c;rport From: "atec" <sip:004970317031735880@localhost;user=phone>;tag=31943 To: <sip:00491721009776@217.110.34.74;user=phone>;tag=gK0290f8b5 Call-ID: 9d46514b@pbx CSeq: 10383 ACK Max-Forwards: 70 Contact: <sip:703173588@10.10.0.11:5060;transport=udp> Content-Length: 0 [5] 2012/03/30 12:35:12: INVITE Response 487 Request Terminated: Terminate 9d46514b@pbx [7] 2012/03/30 12:35:12: 3c267bc0976c-ccg452e9pmln: Media-aware pass-through mode [8] 2012/03/30 12:35:12: Clearing call port 63, SIP call id 9d46514b@pbx [8] 2012/03/30 12:35:12: Remove leg 51: call port 63, SIP call id 9d46514b@pbx [7] 2012/03/30 12:35:12: SIP Rx tls:10.10.0.49:2059: ACK sip:01721009776@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 10.10.0.49:2059;branch=z9hG4bK-cvmxx4xhthyu;rport From: "Zentrale atec" <sip:49@localhost>;tag=xmyfg9zmlg To: <sip:01721009776@localhost;user=phone>;tag=b8c4bec1ec Call-ID: 3c267bc0976c-ccg452e9pmln CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:49@10.10.0.49:2059;transport=tls;line=yoxynt0d>;reg-id=1 Proxy-Require: buttons Content-Length: 0 [8] 2012/03/30 12:35:12: Packet authenticated by transport layer [8] 2012/03/30 12:35:12: Hangup: Call 62 not found [8] 2012/03/30 12:35:12: Clearing call port 62, SIP call id 3c267bc0976c-ccg452e9pmln

Hallo Zusammen, ich habe ein Problem mit einem SIP Trunk von der Firma Colt Telecom. Diese haben mit nur eine IP mitgegeben und das soll dann wohl direkt auf deeren Border Controller laufen (heißt das so) ich bekomme das allerdings nicht in der Snom konfiguriert. Einstellungen als Gateway mit der IP als SIP Proxy bereits getrestet, bekomme aber immer die Fehlermeldung: Dialplan "Standard Schema": Match 01721009776@localhost to <sip:01721009776@217.110.34.74;user=phone> on trunk Colt [5] 2012/03/30 11:29:27: INVITE Response 403 Forbidden: Terminate 51988a3d@pbx weiß jemand eine Lösung?