Tom Tom
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Posts posted by Tom Tom
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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.
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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.
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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
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Telefonieren rein und raus geht.. 1. Fehler bei Colt, falsche IP hinterlegt.
2. Fehler Port im Router nicht freigegeben.
Danke für die Hilfe
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Mittlerweile hat der Provider nachgebessert jetzt gehen die Anrufe wenigtens raus aber es kommt kein Ruf an.
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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?
Cisco ATA186 konfig für SnomOne?
in German
Posted
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