606 Not Acceptable Telekom VOIP bei einer Nummer

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T

theneon

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Hallo zusammen ,

bei einerm Nummer hab ich ein Problem. Ist die Nummer von einem bekannten .
Kommt immer 606 ? Was kann das sein ?

Meine Codecs aktuell : allow=!all,g722,ulaw,alaw

Code: 
<--- SIP read from UDP:217.0.27.68:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP x.x.x.x:5160;received=91.47.139.138;rport=5160;branch=z9hG4bK6c6e92d5
To: <sip:[email protected]>;tag=h7g4Esbg_p65549t1506016888m66908c214134343s1_3788869392-1
From: "0111111111" <sip:[email protected]>;tag=as55e489ac
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.27.68;transport=udp;lr>
P-Early-Media: sendonly
Supported: timer
Content-Type: application/sdp
Content-Length: 176
Allow: UPDATE, NOTIFY, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Accept: application/media_control+xml
Accept: application/vnd.etsi.cug+xml

v=0
o=- 1513793350 3788869378 IN IP4 217.0.27.68
s=IMSS
c=IN IP4 217.0.4.196
t=0 0
m=audio 19890 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
<------------->
--- (18 headers 9 lines) ---
list_route: hop: <sip:217.0.27.68;transport=udp;lr>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g722), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.0.4.196:19890
    -- SIP/DTAG-IP-000007dd is making progress passing it to SIP/06-000007dc

<--- SIP read from UDP:192.168.2.203:5160 --->
INVITE sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.203:5160;rport;branch=z9hG4bKPjEdT8hEPR8h2eFeFm3EEip1PYt4oDJ7Zq
Max-Forwards: 70
From: "06" <sip:[email protected]>;tag=NyXsyQhrpnOlxx4hBNDWsKYDhWKLHcxM
To: <sip:[email protected]>
Contact: "06" <sip:[email protected]:5160;ob>
Call-ID: zvsAgwyndTu4hEUc5uIcVx-96S0umGKq
CSeq: 32340 INVITE
Route: <sip:x.x.x.x:5160;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.1.7
Authorization: Digest username="06", realm="asterisk", nonce="26df8f2e", uri="sip:[email protected]:5160", response="05d877a5ea19e786f53beca757a701be", algorithm=MD5
Content-Type: application/sdp
Content-Length: 485

v=0
o=- 3715005685 3715005685 IN IP4 192.168.2.203
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 103 102 104 109 3 0 8 9 101
c=IN IP4 192.168.2.203
b=TIAS:64000
a=rtcp:4007 IN IP4 192.168.2.203
a=sendrecv
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (15 headers 22 lines) ---
Ignoring this INVITE request

<--- Transmitting (no NAT) to 192.168.2.203:5160 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.203:5160;branch=z9hG4bKPjEdT8hEPR8h2eFeFm3EEip1PYt4oDJ7Zq;received=192.168.2.203;rport=5160
From: "06" <sip:[email protected]>;tag=NyXsyQhrpnOlxx4hBNDWsKYDhWKLHcxM
To: <sip:[email protected]>
Call-ID: zvsAgwyndTu4hEUc5uIcVx-96S0umGKq
CSeq: 32340 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5160>
Content-Length: 0


<------------>

<--- SIP read from UDP:217.0.27.68:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP x.x.x.x:5160;received=91.47.139.138;rport=5160;branch=z9hG4bK6c6e92d5
To: <sip:[email protected]>;tag=h7g4Esbg_p65549t1506016888m66908c214134343s1_3790160719-1
From: "0111111111" <sip:[email protected]>;tag=as55e489ac
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.27.68;transport=udp;lr>
P-Early-Media: sendonly
Supported: timer
Content-Type: application/sdp
Content-Length: 176
Allow: UPDATE, NOTIFY, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Accept: application/media_control+xml
Accept: application/vnd.etsi.cug+xml

v=0
o=- 1870648190 3790160704 IN IP4 217.0.27.68
s=IMSS
c=IN IP4 217.0.4.196
t=0 0
m=audio 19890 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
<------------->
--- (18 headers 9 lines) ---
list_route: hop: <sip:217.0.27.68;transport=udp;lr>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g722), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.0.4.196:19890
    -- SIP/DTAG-IP-000007dd is making progress passing it to SIP/06-000007dc

<--- SIP read from UDP:217.0.27.68:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP x.x.x.x:5160;received=91.47.139.138;rport=5160;branch=z9hG4bK6c6e92d5
To: <sip:[email protected]>;tag=h7g4Esbg_p65549t1506016888m66908c214134343s1_3792175544-1
From: "0111111111" <sip:[email protected]>;tag=as55e489ac
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.27.68;transport=udp;lr>
P-Early-Media: sendonly
Supported: timer
Content-Type: application/sdp
Content-Length: 176
Allow: UPDATE, NOTIFY, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Accept: application/media_control+xml
Accept: application/vnd.etsi.cug+xml

v=0
o=- 1488041113 3792175529 IN IP4 217.0.27.68
s=IMSS
c=IN IP4 217.0.4.196
t=0 0
m=audio 19890 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
<------------->
--- (18 headers 9 lines) ---
list_route: hop: <sip:217.0.27.68;transport=udp;lr>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g722), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.0.4.196:19890
    -- SIP/DTAG-IP-000007dd is making progress passing it to SIP/06-000007dc

<--- SIP read from UDP:217.0.27.68:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP x.x.x.x:5160;received=91.47.139.138;rport=5160;branch=z9hG4bK6c6e92d5
To: <sip:[email protected]>;tag=h7g4Esbg_p65549t1506016888m66908c214134343s1_3793644531-1
From: "0111111111" <sip:[email protected]>;tag=as55e489ac
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.27.68;transport=udp;lr>
P-Early-Media: sendonly
Supported: timer
Content-Type: application/sdp
Content-Length: 176
Allow: UPDATE, NOTIFY, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Accept: application/media_control+xml
Accept: application/vnd.etsi.cug+xml

v=0
o=- 2007655718 3793644516 IN IP4 217.0.27.68
s=IMSS
c=IN IP4 217.0.4.196
t=0 0
m=audio 19890 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
<------------->
--- (18 headers 9 lines) ---
list_route: hop: <sip:217.0.27.68;transport=udp;lr>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g722), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.0.4.196:19890
    -- SIP/DTAG-IP-000007dd is making progress passing it to SIP/06-000007dc

<--- SIP read from UDP:192.168.2.87:5060 --->


<------------->

<--- SIP read from UDP:217.0.27.68:5060 --->
SIP/2.0 606 Not Acceptable
Via: SIP/2.0/UDP x.x.x.x:5160;received=91.47.139.138;rport=5160;branch=z9hG4bK6c6e92d5
To: <sip:[email protected]>;tag=h7g4Esbg_p65549t1506016888m66908c214134343s1_3785594778-1850539287
From: "0111111111" <sip:[email protected]>;tag=as55e489ac
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];transport=udp>
Reason: Q.850;cause=88;text="5"
Supported: timer
Content-Length: 0
 
Zuletzt bearbeitet von einem Moderator:
mmh wird da eine Rufnummer übertragen beim ankommenden Gespräch ?
wenn keine Nr. übertragen wird solltest du mal nachsehen ob (anonyme Call allow) aktiviert ist ! (Sip Konfig)
kann keine cid sehen ...
 
Da ich die Vorgehensweisen des neuen Forum-Betreibers nicht gut heiße, stehen meine Beiträge hier nicht weiter zur Verfügung.
 
Zuletzt bearbeitet von einem Moderator:
Um antworten zu können musst du eingeloggt sein.

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