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