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Wahl der Nummer vom System verweigert ...

samy77

By samy77
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samy77 Newbie

samy77

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Posted

Hallo,

 

habe versucht eine Weiterleitung für alle Nummern über einen Ein/Ausschalter zu machen.

Leider bekomme ich beim Aufrufen vom Telefon aus diese Ansage:

 

Die Wahl dieser Nummer wurde vom System verweigert !

 

Der Angemeldete User auf dem Telefon hat sogar Admin Rechte daher bin ich ratlos was das sein könnte.

 

Hier wäre ein Auszug vom Log von diesem Vorgang:

"

[5] 2012/03/31 10:17:39: SIP Rx tls:192.168.1.21:3520:

INVITE sip:74@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.21:3520;branch=z9hG4bK-lk1g8hzywlp6;rport

From: "USER" <sip:11@localhost>;tag=hy8jssh5ie

To: <sip:74@localhost;user=phone>

Call-ID: 3c39aed015f0-1evwgmeulfri

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:11@192.168.1.21:3520;transport=tls;line=p8o149c9>;reg-id=1

X-Serialnumber: 0004133A2207

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

User-Agent: snom370/8.4.18

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: 524

 

v=0

o=root 1756067162 1756067162 IN IP4 192.168.1.21

s=call

c=IN IP4 192.168.1.21

t=0 0

m=audio 54250 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:GtzdjoS2aJopzMkbhxzAZ2pDwaMu7PuSDqV817DD

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] 2012/03/31 10:17:39: Packet authenticated by transport layer

[8] 2012/03/31 10:17:39: Allocating for call port 49, SIP call id 3c39aed015f0-1evwgmeulfri

[9] 2012/03/31 10:17:39: UDP(IPv4): Opening socket on 0.0.0.0:52850

[9] 2012/03/31 10:17:39: UDP(IPv4): Opening socket on 0.0.0.0:52851

[9] 2012/03/31 10:17:39: UDP(IPv6): Opening socket on [::]:52850

[9] 2012/03/31 10:17:39: UDP(IPv6): Opening socket on [::]:52851

[8] 2012/03/31 10:17:39: Could not find a trunk (1 trunks)

[9] 2012/03/31 10:17:39: Using outbound proxy sip:192.168.1.21:3520;transport=tls because of flow-label

[9] 2012/03/31 10:17:39: Last message repeated 3 times

[5] 2012/03/31 10:17:39: SIP Tx tls:192.168.1.21:3520:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.1.21:3520;branch=z9hG4bK-lk1g8hzywlp6;rport=3520

From: "USER" <sip:11@localhost>;tag=hy8jssh5ie

To: <sip:74@localhost;user=phone>;tag=c0f86c8669

Call-ID: 3c39aed015f0-1evwgmeulfri

CSeq: 1 INVITE

Content-Length: 0

 

[7] 2012/03/31 10:17:39: Set packet length to 20

[6] 2012/03/31 10:17:39: Call-leg 49: Sending RTP for 3c39aed015f0-1evwgmeulfri to 192.168.1.21:54250, codec not set yet

[8] 2012/03/31 10:17:39: Incoming call: Request URI sip:74@localhost;user=phone, To is <sip:74@localhost;user=phone>

[8] 2012/03/31 10:17:39: Call from an user 11

[8] 2012/03/31 10:17:39: To is <sip:74@localhost;user=phone>, user 28, domain 1

[8] 2012/03/31 10:17:39: To user 74

[8] 2012/03/31 10:17:39: Set the To domain based on From user 11@localhost

[8] 2012/03/31 10:17:39: Call state for call object 19: idle

[5] 2012/03/31 10:17:39: Sending IM from "USER" <sip:11@localhost> to "USER" <sip:11@localhost> (1 destinations)

[5] 2012/03/31 10:17:39: Last message repeated 2 times

[8] 2012/03/31 10:17:39: Call state for call object 19: connected

[5] 2012/03/31 10:17:39: Sending IM from "USER" <sip:11@localhost> to "USER" <sip:11@localhost> (1 destinations)

[5] 2012/03/31 10:17:39: Last message repeated 2 times

[8] 2012/03/31 10:17:39: Play audio_de/ex_permission.wav, caching false

[7] 2012/03/31 10:17:39: Call port 49: set_codecs for 3c39aed015f0-1evwgmeulfri codecs "", codec_preference count 6

[9] 2012/03/31 10:17:39: Using outbound proxy sip:192.168.1.21:3520;transport=tls because of flow-label

[9] 2012/03/31 10:17:39: Last message repeated 4 times

[7] 2012/03/31 10:17:39: Set packet length to 20

[9] 2012/03/31 10:17:39: Call port 49: update_codecs for 3c39aed015f0-1evwgmeulfri: adding codec pcmu/8000 to available list

[9] 2012/03/31 10:17:39: Call port 49: update_codecs for 3c39aed015f0-1evwgmeulfri: adding codec pcma/8000 to available list

[9] 2012/03/31 10:17:39: Call port 49: update_codecs for 3c39aed015f0-1evwgmeulfri: adding codec g722/8000 to available list

[9] 2012/03/31 10:17:39: Call port 49: update_codecs for 3c39aed015f0-1evwgmeulfri: adding codec g726-32/8000 to available list

[9] 2012/03/31 10:17:39: Call port 49: update_codecs for 3c39aed015f0-1evwgmeulfri: adding codec gsm/8000 to available list

[9] 2012/03/31 10:17:39: Call port 49: update_codecs for 3c39aed015f0-1evwgmeulfri: codec_preference size 6, available codecs size 6

[5] 2012/03/31 10:17:39: set codec: codec pcmu/8000 is set to call-leg 49

[6] 2012/03/31 10:17:39: Call-leg 49: Codec pcmu/8000 is chosen for call id 3c39aed015f0-1evwgmeulfri

[5] 2012/03/31 10:17:39: SIP Tx tls:192.168.1.21:3520:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.21:3520;branch=z9hG4bK-lk1g8hzywlp6;rport=3520

From: "USER" <sip:11@localhost>;tag=hy8jssh5ie

To: <sip:74@localhost;user=phone>;tag=c0f86c8669

Call-ID: 3c39aed015f0-1evwgmeulfri

CSeq: 1 INVITE

Contact: <sip:11@192.168.1.248:5061;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/2011-4.5.0.1016 Alpha Monocerotids

Content-Type: application/sdp

Content-Length: 433

 

v=0

o=- 346625246 346625246 IN IP4 192.168.1.248

s=-

c=IN IP4 192.168.1.248

t=0 0

m=audio 52850 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:9vKc6GMuBPPEeTtK050QYw1egZlpilqtLOKJnzhV

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

[5] 2012/03/31 10:17:39: set codec: codec pcmu/8000 is set to call-leg 49

[6] 2012/03/31 10:17:39: Call-leg 49: Codec pcmu/8000 is chosen for call id 3c39aed015f0-1evwgmeulfri

[5] 2012/03/31 10:17:40: SIP Rx tls:192.168.1.21:3520:

ACK sip:11@192.168.1.248:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.21:3520;branch=z9hG4bK-1ch4xlbh1lrh;rport

From: "USER" <sip:11@localhost>;tag=hy8jssh5ie

To: <sip:74@localhost;user=phone>;tag=c0f86c8669

Call-ID: 3c39aed015f0-1evwgmeulfri

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:11@192.168.1.21:3520;transport=tls;line=p8o149c9>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

[8] 2012/03/31 10:17:40: Packet authenticated by transport layer

[8] 2012/03/31 10:17:40: SRTP MAC mismatch: 10516b96 != 4f4d0000

[7] 2012/03/31 10:17:40: Discard SRTCP packet from 192.168.1.21:54251 with wrong MAC

[5] 2012/03/31 10:17:42: SMTP: No email server specified

[9] 2012/03/31 10:17:43: Resolve 173020: aaaa udp 213.164.25.150 5060

[9] 2012/03/31 10:17:43: Resolve 173020: a udp 213.164.25.150 5060

[9] 2012/03/31 10:17:43: Resolve 173020: udp 213.164.25.150 5060

[5] 2012/03/31 10:17:43: SIP Tx tls:192.168.1.21:3520:

BYE sip:11@192.168.1.21:3520;transport=tls;line=p8o149c9 SIP/2.0

Via: SIP/2.0/TLS 192.168.1.248:5061;branch=z9hG4bK-807fb2e3168d1aa695a0d14ad4b87606;rport

From: <sip:74@localhost;user=phone>;tag=c0f86c8669

To: "USER" <sip:11@localhost>;tag=hy8jssh5ie

Call-ID: 3c39aed015f0-1evwgmeulfri

CSeq: 20897 BYE

Max-Forwards: 70

Contact: <sip:11@192.168.1.248:5061;transport=tls>

Content-Length: 0

 

[5] 2012/03/31 10:17:43: SIP Rx tls:192.168.1.21:3520:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 192.168.1.248:5061;branch=z9hG4bK-807fb2e3168d1aa695a0d14ad4b87606;rport=5061

From: <sip:74@localhost;user=phone>;tag=c0f86c8669

To: "USER" <sip:11@localhost>;tag=hy8jssh5ie

Call-ID: 3c39aed015f0-1evwgmeulfri

CSeq: 20897 BYE

Contact: <sip:11@192.168.1.21:3520;transport=tls;line=p8o149c9>;reg-id=1

User-Agent: snom370/8.4.18

RTP-RxStat: Total_Rx_Pkts=204,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=179,Tx_Pkts=179,Remote_Tx_Pkts=5

Content-Length: 0

 

[7] 2012/03/31 10:17:43: Call 3c39aed015f0-1evwgmeulfri: Clear last request

[5] 2012/03/31 10:17:43: BYE Response: Terminate 3c39aed015f0-1evwgmeulfri

[8] 2012/03/31 10:17:43: Remove leg 22: call port 49, SIP call id 3c39aed015f0-1evwgmeulfri

[8] 2012/03/31 10:17:43: Clearing call port 49, SIP call id 3c39aed015f0-1evwgmeulfri

[5] 2012/03/31 10:17:43: Sending IM from "USER" <sip:11@localhost> to "USER" <sip:11@localhost> (1 destinations)

[5] 2012/03/31 10:17:43: Last message repeated 2 times

[9] 2012/03/31 10:17:43: Using outbound proxy sip:192.168.1.21:3520;transport=tls because of flow-label"

 

 

 

Hat dazu jemand eine Idee was ich ändern muss?

 

Thx

Samy

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