[Problem] HANGUPCAUSE = 58 bei Telekom Trunk

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davidb

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Hallo zusammen,

nach ewigem googlen hoffe ich, dass mir von euch jemand weiterhelfen kann. Folgende Situation:

Ich habe ein Telekom VDSL. Der Speedport von der DTAG wird nur als Modem genutzt. Dahinter hängt ein PFSense mit der PPPoE-Einwahl. Mein Asterisk (mit FreePBX) hängt dahinter.

Meine Trunks zur Telekom connecten. Eingehende Anrufe funktionieren ohne Probleme. Ausgehende Anrufe über einen Telekom-Trunk brechen aber mit der Meldung "DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 58" ab. Über einen Vergleichs-Trunk zu Sipgate funktioniert alles. Wenn ich das recht deute, fordert mein Asterisk bei der Verbindung eine Funktion, die die Telekom nicht unterstützt, oder? Hat eventuell jemand einen Tip für mich?

Meine Trunk-Config habe ich von hier übernommen: http://gergernaut.de/telekom-voip-mit-asterisk-freepbx-als-sip-trunk/

Die Peer Details:
Code: 
type=friend
username=RUFNUMMER
fromuser=RUFNUMMER
secret=PASSWORT
host=tel.t-online.de
nat=yes
dtmfmode=rfc2833
canreinvite=update
fromdomain=tel.t-online.de
insecure=very
qualify=yes

Unten noch ein ausführliches Log von einem versuchten Anruf.

Vorab vielen Dank für jede Hilfe und viele Grüße,

David

Code: 
<--- SIP read from UDP:192.168.102.20:5060 --->
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bK5256dbbb2b20edec9
Max-Forwards: 70
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>
Call-ID: aafa9439a1fdb369
CSeq: 20924411 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE
Allow-Events: hold
Contact: "TELSCHLAF01" <sip:[email protected]:5060;transport=udp>
Supported: replaces, 100rel
User-Agent: OpenStage_60_V3 R1.31.0 SIP 120727
X-Siemens-Call-Type: ST-insecure
Content-Type: application/sdp
Content-Length: 404

v=0
o=OpenStage-Line_1 1361213574 1029310706 IN IP4 192.168.102.20
s=SIP Call
c=IN IP4 192.168.102.20
t=0 0
m=audio 5010 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
m=video 5050 RTP/AVP 34
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1
a=recvonly
<------------->
--- (15 headers 18 lines) ---
Sending to 192.168.102.20:5060 (NAT)
Sending to 192.168.102.20:5060 (NAT)
Using INVITE request as basis request - aafa9439a1fdb369
Found peer '201' for '201' from 192.168.102.20:5060

<--- Reliably Transmitting (NAT) to 192.168.102.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bK5256dbbb2b20edec9;received=192.168.102.20;rport=5060
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>;tag=as38edb6ce
Call-ID: aafa9439a1fdb369
CSeq: 20924411 INVITE
Server: FPBX-12.0.76(11.12.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1597c5bd"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'aafa9439a1fdb369' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.102.20:5060 --->
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bK5256dbbb2b20edec9
Max-Forwards: 70
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>;tag=as38edb6ce
Call-ID: aafa9439a1fdb369
CSeq: 20924411 ACK
User-Agent: OpenStage_60_V3 R1.31.0 SIP 120727
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.102.20:5060 --->
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bKea0b33fff590d8ca6
Max-Forwards: 70
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>
Call-ID: aafa9439a1fdb369
CSeq: 20924412 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, UPDATE
Allow-Events: hold
Authorization: Digest username="201",realm="asterisk",nonce="1597c5bd",uri="sip:[email protected]:5060;transport=udp",response="11df52c20639c43f31714f23479c6e0a",algorithm=MD5
Contact: "TELSCHLAF01" <sip:[email protected]:5060;transport=udp>
Supported: replaces, 100rel
User-Agent: OpenStage_60_V3 R1.31.0 SIP 120727
X-Siemens-Call-Type: ST-insecure
Content-Type: application/sdp
Content-Length: 404

v=0
o=OpenStage-Line_1 1361213574 1029310706 IN IP4 192.168.102.20
s=SIP Call
c=IN IP4 192.168.102.20
t=0 0
m=audio 5010 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
m=video 5050 RTP/AVP 34
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1
a=recvonly
<------------->
--- (16 headers 18 lines) ---
Sending to 192.168.102.20:5060 (NAT)
Using INVITE request as basis request - aafa9439a1fdb369
Found peer '201' for '201' from 192.168.102.20:5060
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - (g723|gsm|ulaw|alaw|g726|g722|h263|h263p|h264|mpeg4), peer - audio=(ulaw|alaw|g729)/video=(h263)/text=(nothing), combined - (ulaw|alaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.102.20:5010
Peer video RTP is at port 192.168.102.20:5050
Looking for 00176ZZZZZZZZ in from-internal (domain 192.168.102.1)
list_route: hop: <sip:[email protected]:5060;transport=udp>
       > [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('CHAN_START',{ts '2015-10-11 19:17:30'},'Schlafzimmer - Privat','201','','','','00176ZZZZZZZZ','from-internal','SIP/201-00000004','','',3,'','1444583850.4','1444583850.4','','','')]

<--- Transmitting (NAT) to 192.168.102.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bKea0b33fff590d8ca6;received=192.168.102.20;rport=5060
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>
Call-ID: aafa9439a1fdb369
CSeq: 20924412 INVITE
Server: FPBX-12.0.76(11.12.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Executing [00176ZZZZZZZZ@from-internal:1] Macro("SIP/201-00000004", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/201-00000004", "TOUCH_MONITOR=1444583850.4") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/201-00000004", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/201-00000004", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/201-00000004", "1?Set(REALCALLERIDNUM=201)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/201-00000004", "AMPUSER=201") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/201-00000004", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/201-00000004", "AMPUSERCIDNAME=Schlafzimmer - Privat") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/201-00000004", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/201-00000004", "AMPUSERCID=201") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/201-00000004", "__DIAL_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/201-00000004", "CALLERID(all)="Schlafzimmer - Privat" <201>") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/201-00000004", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("SIP/201-00000004", "1?Set(GROUP(concurrency_limit)=201)") in new stack
    -- Executing [s@macro-user-callerid:14] GosubIf("SIP/201-00000004", "7?sub-ccss,s,1(from-internal,00176ZZZZZZZZ)") in new stack
    -- Executing [s@sub-ccss:1] ExecIf("SIP/201-00000004", "0?Return()") in new stack
    -- Executing [s@sub-ccss:2] Set("SIP/201-00000004", "CCSS_SETUP=TRUE") in new stack
    -- Executing [s@sub-ccss:3] GosubIf("SIP/201-00000004", "0?monitor_config,1(from-internal,00176ZZZZZZZZ):monitor_default,1(from-internal,00176ZZZZZZZZ)") in new stack
    -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/201-00000004", "0?is_exten") in new stack
    -- Executing [monitor_default@sub-ccss:2] StackPop("SIP/201-00000004", "") in new stack
    -- Executing [monitor_default@sub-ccss:3] Return("SIP/201-00000004", "FALSE") in new stack
    -- Executing [s@macro-user-callerid:15] ExecIf("SIP/201-00000004", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:16] GotoIf("SIP/201-00000004", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,30)
    -- Executing [s@macro-user-callerid:30] Set("SIP/201-00000004", "CALLERID(number)=201") in new stack
    -- Executing [s@macro-user-callerid:31] Set("SIP/201-00000004", "CALLERID(name)=Schlafzimmer - Privat") in new stack
    -- Executing [s@macro-user-callerid:32] Set("SIP/201-00000004", "CDR(cnum)=201") in new stack
    -- Executing [s@macro-user-callerid:33] Set("SIP/201-00000004", "CDR(cnam)=Schlafzimmer - Privat") in new stack
    -- Executing [s@macro-user-callerid:34] Set("SIP/201-00000004", "CHANNEL(language)=en") in new stack
    -- Executing [00176ZZZZZZZZ@from-internal:2] Gosub("SIP/201-00000004", "sub-record-check,s,1(out,00176ZZZZZZZZ,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("SIP/201-00000004", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("SIP/201-00000004", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("SIP/201-00000004", "NOW=1444583850") in new stack
    -- Executing [s@sub-record-check:4] Set("SIP/201-00000004", "__DAY=11") in new stack
    -- Executing [s@sub-record-check:5] Set("SIP/201-00000004", "__MONTH=10") in new stack
    -- Executing [s@sub-record-check:6] Set("SIP/201-00000004", "__YEAR=2015") in new stack
    -- Executing [s@sub-record-check:7] Set("SIP/201-00000004", "__TIMESTR=20151011-191730") in new stack
    -- Executing [s@sub-record-check:8] Set("SIP/201-00000004", "__FROMEXTEN=201") in new stack
    -- Executing [s@sub-record-check:9] Set("SIP/201-00000004", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("SIP/201-00000004", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("SIP/201-00000004", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("SIP/201-00000004", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("SIP/201-00000004", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("SIP/201-00000004", "3?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("SIP/201-00000004", "1?sub-record-check,out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] NoOp("SIP/201-00000004", "Outbound Recording Check from 201 to 00176ZZZZZZZZ") in new stack
    -- Executing [out@sub-record-check:2] Set("SIP/201-00000004", "RECMODE=dontcare") in new stack
    -- Executing [out@sub-record-check:3] ExecIf("SIP/201-00000004", "1?Goto(routewins)") in new stack
    -- Goto (sub-record-check,out,7)
    -- Executing [out@sub-record-check:7] Gosub("SIP/201-00000004", "recordcheck,1(dontcare,out,00176ZZZZZZZZ)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/201-00000004", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("SIP/201-00000004", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("SIP/201-00000004", "") in new stack
    -- Executing [out@sub-record-check:8] Return("SIP/201-00000004", "") in new stack
    -- Executing [00176ZZZZZZZZ@from-internal:3] ExecIf("SIP/201-00000004", "0 ?Set(CDR(accountcode)=)") in new stack
    -- Executing [00176ZZZZZZZZ@from-internal:4] Set("SIP/201-00000004", "MOHCLASS=default") in new stack
    -- Executing [00176ZZZZZZZZ@from-internal:5] Set("SIP/201-00000004", "_NODEST=") in new stack
    -- Executing [00176ZZZZZZZZ@from-internal:6] Macro("SIP/201-00000004", "dialout-trunk,2,0176ZZZZZZZZ,,off") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/201-00000004", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/201-00000004", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/201-00000004", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/201-00000004", "DIAL_NUMBER=0176ZZZZZZZZ") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/201-00000004", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/201-00000004", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/201-00000004", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/201-00000004", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/201-00000004", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/201-00000004", "DIAL_TRUNK_OPTIONS=Tt") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/201-00000004", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/201-00000004", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/201-00000004", "0?Set(REALCALLERIDNUM=201)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/201-00000004", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/201-00000004", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/201-00000004", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/201-00000004", "TRUNKOUTCID=0661SSSSSSSS") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/201-00000004", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,14)
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/201-00000004", "1?Set(CALLERID(all)=0661SSSSSSSS)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/201-00000004", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/201-00000004", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/201-00000004", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:18] Set("SIP/201-00000004", "CDR(outbound_cnum)=0661SSSSSSSS") in new stack
    -- Executing [s@macro-outbound-callerid:19] Set("SIP/201-00000004", "CDR(outbound_cnam)=") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/201-00000004", "0?sub-flp-2,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/201-00000004", "OUTNUM=0176ZZZZZZZZ") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/201-00000004", "custom=SIP/0661SSSSSSSS") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/201-00000004", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/201-00000004", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/201-00000004", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/201-00000004", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/201-00000004", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/201-00000004", "1?Set(CONNECTEDLINE(num,i)=0176ZZZZZZZZ)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/201-00000004", "1?Set(CONNECTEDLINE(name,i)=CID:0661SSSSSSSS)") in new stack
    -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/201-00000004", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/201-00000004", "SIP/0661SSSSSSSS/0176ZZZZZZZZ,300,Tt") in new stack
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 30604
Video is at 192.168.102.1:30074
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100012 (g722) to SDP
Adding video codec 200004 (h264) to SDP
Adding video codec 200005 (mpeg4) to SDP
Adding video codec 200003 (h263p) to SDP
Adding video codec 200002 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 217.0.23.100:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.102.1:5060;branch=z9hG4bK668e5167;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as478ee4e4
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76(11.12.0)
Date: Sun, 11 Oct 2015 17:17:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 743

v=0
o=root 1009606198 1009606198 IN IP4 192.168.102.1
s=Asterisk PBX 11.12.0
c=IN IP4 192.168.102.1
b=CT:4096
t=0 0
m=audio 30604 RTP/AVP 8 0 111 3 4 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30074 RTP/AVP 99 104 98 34
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

---
    -- Called SIP/0661SSSSSSSS/0176ZZZZZZZZ
       > [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('CHAN_START',{ts '2015-10-11 19:17:30'},'','','','','','s','from-trunk-sip-0661SSSSSSSS','SIP/0661SSSSSSSS-00000005','','',3,'','1444583850.5','1444583850.4','','','')]
Retransmitting #1 (NAT) to 217.0.23.100:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.102.1:5060;branch=z9hG4bK668e5167;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as478ee4e4
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-12.0.76(11.12.0)
Date: Sun, 11 Oct 2015 17:17:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 743

v=0
o=root 1009606198 1009606198 IN IP4 192.168.102.1
s=Asterisk PBX 11.12.0
c=IN IP4 192.168.102.1
b=CT:4096
t=0 0
m=audio 30604 RTP/AVP 8 0 111 3 4 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30074 RTP/AVP 99 104 98 34
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

---

<--- SIP read from UDP:217.0.23.100:5060 --->
SIP/2.0 407 Proxy Authentication Required 020350304
Via: SIP/2.0/UDP 192.168.102.1:5060;received=87.145.33.61;rport=24766;branch=z9hG4bK668e5167
To: <sip:[email protected]>;tag=h7g4Esbg_499b72360344e30050bd6451323e273c
From: <sip:[email protected]>;tag=as478ee4e4
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce="80EC400FB5991A5600000000265DEA73",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 217.0.23.100:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.102.1:5060;branch=z9hG4bK668e5167;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as478ee4e4
To: <sip:[email protected]>;tag=h7g4Esbg_499b72360344e30050bd6451323e273c
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: FPBX-12.0.76(11.12.0)
Content-Length: 0


---
Audio is at 30604
Video is at 192.168.102.1:30074
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100012 (g722) to SDP
Adding video codec 200004 (h264) to SDP
Adding video codec 200005 (mpeg4) to SDP
Adding video codec 200003 (h263p) to SDP
Adding video codec 200002 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 217.0.23.100:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.102.1:5060;branch=z9hG4bK7035137b;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as478ee4e4
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: FPBX-12.0.76(11.12.0)
Proxy-Authorization: Digest username="0661SSSSSSSS", realm="tel.t-online.de", algorithm=MD5, uri="sip:[email protected]", nonce="80EC400FB5991A5600000000265DEA73", response="9937fb8a6a17d2389db267cab2a27fe9", qop=auth, cnonce="1423955e", nc=00000001
Date: Sun, 11 Oct 2015 17:17:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 743

v=0
o=root 1009606198 1009606199 IN IP4 192.168.102.1
s=Asterisk PBX 11.12.0
c=IN IP4 192.168.102.1
b=CT:4096
t=0 0
m=audio 30604 RTP/AVP 8 0 111 3 4 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30074 RTP/AVP 99 104 98 34
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

---
Retransmitting #1 (NAT) to 217.0.23.100:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.102.1:5060;branch=z9hG4bK7035137b;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as478ee4e4
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: FPBX-12.0.76(11.12.0)
Proxy-Authorization: Digest username="0661SSSSSSSS", realm="tel.t-online.de", algorithm=MD5, uri="sip:[email protected]", nonce="80EC400FB5991A5600000000265DEA73", response="9937fb8a6a17d2389db267cab2a27fe9", qop=auth, cnonce="1423955e", nc=00000001
Date: Sun, 11 Oct 2015 17:17:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 743

v=0
o=root 1009606198 1009606199 IN IP4 192.168.102.1
s=Asterisk PBX 11.12.0
c=IN IP4 192.168.102.1
b=CT:4096
t=0 0
m=audio 30604 RTP/AVP 8 0 111 3 4 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30074 RTP/AVP 99 104 98 34
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

---

<--- SIP read from UDP:217.0.23.100:5060 --->
SIP/2.0 606 Not Acceptable
Via: SIP/2.0/UDP 192.168.102.1:5060;received=87.145.33.61;rport=24766;branch=z9hG4bK7035137b
To: <sip:[email protected]>;tag=h7g4Esbg_p65545t1444583850m943492c483378826s1_1665706597-1201760667
From: <sip:[email protected]>;tag=as478ee4e4
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];transport=udp>
Supported: timer
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 217.0.23.100:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.102.1:5060;branch=z9hG4bK7035137b;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as478ee4e4
To: <sip:[email protected]>;tag=h7g4Esbg_p65545t1444583850m943492c483378826s1_1665706597-1201760667
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: FPBX-12.0.76(11.12.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/201-00000004", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 58") in new stack
    -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/201-00000004", "0?continue,1:s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/201-00000004", "RC=58") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/201-00000004", "58,1") in new stack
    -- Goto (macro-dialout-trunk,58,1)
    -- Executing [58@macro-dialout-trunk:1] Goto("SIP/201-00000004", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/201-00000004", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 58 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:2] Set("SIP/201-00000004", "CALLERID(number)=201") in new stack
    -- Executing [00176ZZZZZZZZ@from-internal:7] Macro("SIP/201-00000004", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/201-00000004", "") in new stack
Audio is at 30782
Video is at 192.168.102.1:30272
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding video codec 200002 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 192.168.102.20:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bKea0b33fff590d8ca6;received=192.168.102.20;rport=5060
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>;tag=as3be12d1b
Call-ID: aafa9439a1fdb369
CSeq: 20924412 INVITE
Server: FPBX-12.0.76(11.12.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 385

v=0
o=root 73269030 73269030 IN IP4 192.168.102.1
s=Asterisk PBX 11.12.0
c=IN IP4 192.168.102.1
b=CT:4096
t=0 0
m=audio 30782 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 30272 RTP/AVP 34
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=sendrecv

<------------>
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/201-00000004", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/201-00000004", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/201-00000004", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/201-00000004> Playing 'all-circuits-busy-now.alaw' (language 'en')
       > [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('HANGUP',{ts '2015-10-11 19:17:31'},'CID:0661SSSSSSSS','00176ZZZZZZZZ','','','','00176ZZZZZZZZ','from-trunk-sip-0661SSSSSSSS','SIP/0661SSSSSSSS-00000005','AppDial','(Outgoing Line)',3,'','1444583850.5','1444583850.4','','','')]
       > [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('CHAN_END',{ts '2015-10-11 19:17:31'},'CID:0661SSSSSSSS','00176ZZZZZZZZ','','','','00176ZZZZZZZZ','from-trunk-sip-0661SSSSSSSS','SIP/0661SSSSSSSS-00000005','AppDial','(Outgoing Line)',3,'','1444583850.5','1444583850.4','','','')]

<--- SIP read from UDP:192.168.102.20:5060 --->
CANCEL sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bKea0b33fff590d8ca6
Max-Forwards: 70
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>
Call-ID: aafa9439a1fdb369
CSeq: 20924412 CANCEL
User-Agent: OpenStage_60_V3 R1.31.0 SIP 120727
Warning: 399 192.168.102.20 "Call is terminated"
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.102.20:5060 (NAT)

<--- Reliably Transmitting (NAT) to 192.168.102.20:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bKea0b33fff590d8ca6;received=192.168.102.20;rport=5060
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>;tag=as3be12d1b
Call-ID: aafa9439a1fdb369
CSeq: 20924412 INVITE
Server: FPBX-12.0.76(11.12.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (NAT) to 192.168.102.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bKea0b33fff590d8ca6;received=192.168.102.20;rport=5060
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>;tag=as3be12d1b
Call-ID: aafa9439a1fdb369
CSeq: 20924412 CANCEL
Server: FPBX-12.0.76(11.12.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/201-00000004' in macro 'outisbusy'
  == Spawn extension (from-internal, 00176ZZZZZZZZ, 7) exited non-zero on 'SIP/201-00000004'
    -- Executing [h@from-internal:1] Hangup("SIP/201-00000004", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-00000004'
       > [INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid,cnum,cnam,outbound_cnum) VALUES ({ ts '2015-10-11 19:17:30' },'201','201','00176ZZZZZZZZ','from-internal','SIP/201-00000004','SIP/0661SSSSSSSS-00000005','Playback','all-circuits-busy-now&pls-try-call-later, noanswer',1,0,'NO ANSWER',3,'1444583850.4','201','Schlafzimmer - Privat','0661SSSSSSSS')]
       > [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('HANGUP',{ts '2015-10-11 19:17:31'},'','201','201','','00176ZZZZZZZZ','h','from-internal','SIP/201-00000004','','',3,'','1444583850.4','1444583850.4','','','')]
       > [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('CHAN_END',{ts '2015-10-11 19:17:31'},'','201','201','','00176ZZZZZZZZ','h','from-internal','SIP/201-00000004','','',3,'','1444583850.4','1444583850.4','','','')]
       > [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('LINKEDID_END',{ts '2015-10-11 19:17:31'},'','201','201','','00176ZZZZZZZZ','h','from-internal','SIP/201-00000004','','',3,'','1444583850.4','1444583850.4','','','')]
Retransmitting #1 (NAT) to 192.168.102.20:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bKea0b33fff590d8ca6;received=192.168.102.20;rport=5060
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>;tag=as3be12d1b
Call-ID: aafa9439a1fdb369
CSeq: 20924412 INVITE
Server: FPBX-12.0.76(11.12.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.102.20:5060 --->
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bKea0b33fff590d8ca6
Max-Forwards: 70
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>;tag=as3be12d1b
Call-ID: aafa9439a1fdb369
CSeq: 20924412 ACK
User-Agent: OpenStage_60_V3 R1.31.0 SIP 120727
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'aafa9439a1fdb369' Method: ACK

<--- SIP read from UDP:192.168.102.20:5060 --->
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.102.20;branch=z9hG4bKea0b33fff590d8ca6
Max-Forwards: 70
From: "TELSCHLAF01" <sip:[email protected]:5060>;tag=6e925080b8;epid=SC2d697a
To: <sip:[email protected]:5060>;tag=as3be12d1b
Call-ID: aafa9439a1fdb369
CSeq: 20924412 ACK
User-Agent: OpenStage_60_V3 R1.31.0 SIP 120727
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SRVPBX01*CLI>
 
Hmm, bei dem "606 Not Acceptable" welcher da auf dein INVITE kommt von der Telekom, würde ich mal ins Blaue tippen, dass die mit dem Videoangebot im sdp nicht klar kommen, und daher verstört mit 606 antworten. k.A. wo/wie man in der FreePBX GUI Video für sip abstellt, aber in der sip.conf ist es "videosupport=no". Stell dann nochmal den Log hier rein.
 
Vielen Dank. Das war der Hinweis, den ich gebraucht habe. Den Videosupport kann ich nicht vollständig abschalten, da ich ihn für Türstationen nutzen möchte. Aber ich habe es jetzt mal mit einem Telefon ohne Video-Unterstützung getestet. Da geht alles ohne Probleme. Interessant ist aber, dass es bei einem Softphone mit aktivierten Video-Codecs auch funktioniert. Es scheint also an den etwas älteren OpenStage Telefonen zu liegen, die ich hier habe. Ich werde mir mal ein anderes Video-Telefon besorgen und damit testen. Wenn es damit auch geht, werde ich mir das bei den OpenStage noch mal genauer anschauen.

Vielen Dank und viele Grüße,

David
 
Gut, wenn es das also tatsächlich war, dann kann du die Videounterstützung feiner eingrenzen, indem du Video im context des telekom Peer deaktivierst.
So z.B. nur g711 zulassen:

Code: 
disallow=all
allow=alaw
allow=ulaw
 
Um antworten zu können musst du eingeloggt sein.

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