[Gelöst] Anrufweiterleitung zu externen Teilnehmer sendet keine RTP Packet / Kein Ton

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vwittich

vwittich

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Hallo zusammen,

ich habe mal wieder ein Problem, das ich trotz intensiven testen nicht in den Griff bekomme. Ich habe eine Extention eingefügt die alle eingehenden Anrufe weiterleiten. Mit der Extention 44(+Rufnummer) können so alle Anrufe direkt ohne Annahme weitergeleitet werden. Das Problem ist nur, dass die Audioweiterleitung nicht funktioniert. Beim debugen der RTP Packete wird dies deutlich... es findet einfach keine Übertragung statt.

Um das Problem mal nachzuvollziehen, hier meine Konfiguration:
[sip.conf]
Code: 
[general]
context=sonstige   
allowoverlap=no 
udpbindaddr=0.0.0.0:5060
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
language=de   

allowsubscribe=yes
notifyringing = yes
notifyhold = yes
defaultexpirey=240
directmedia=no                  ; das ehemalige canreinvite
alwaysauthreject=yes      
allowguest=no              

register => 063333333:passwort:[email protected]/063333333
register => sip-telmy.com-000000:[email protected]/000000

[...]

[telekom]
type=friend
context=von_telekom
username=username
secret=passwort
host=tel.t-online.de
fromdomain=tel.t-online.de
qualify=yes           
port=5060
insecure=invite
nat=yes
call-limit=5
canreinvite=no

[...]

[telmy]
type=friend
context=von_telmy
username=sip-telmy.com-000000
fromuser=sip-telmy.com-000000
secret=passwort
host=sip.telmy.com
fromdomain=sip.telmy.com
qualify=yes        
port=5060
insecure=invite
nat=yes
call-limit=5
canreinvite=no

Hier mein Dialplan in Auszügen:
[extention.conf]
Code: 
[sonstige]

[globals]
KLINGELZEIT=20

[telefon]
include => call_sip_accounts
include => BLF_group_pickup
include => vorwahl
include => extern
include => sondernr

exten => i,1,NoOp(Undefinierte Nummer ${INVALID_EXTEN} wurde gewaehlt.)
exten => i,2,Answer()
exten => i,3,Playback(that-is-not-rec-phn-num)
exten => i,4,Hangup()

[..]

[extern]
exten => _0049[2-9].,1,set(CALLERID(name)=063333333)
exten => _0049[2-9].,n,set(CALLERID(num)=063333333)
exten => _0049[2-9].,n,Dial(SIP/${EXTEN}@telekom,60,trg)
exten => _0049[2-9].,n,Hangup()

exten => _00ZX.,1,set(CALLERID(name)=4963333333)
exten => _00ZX.,n,set(CALLERID(number)=000000)
exten => _00ZX.,n,Dial(SIP/${EXTEN}@telmy,60,trg)
exten => _00ZX.,n,Hangup()

[...]

[call_sip_accounts]
[...]
exten => _44X.,1,Answer()
exten => _44X.,n,Set(DB(CF/anlage)=${EXTEN:2})
exten => _44X.,n,SayDigits(${EXTEN:2})
exten => _44X.,n,NoOp(Weiterleitung fuer Anlage auf ${EXTEN:2} aktiviert.)
exten => _44X.,n,Hangup()

exten => 44,1,Answer()
exten => 44,n,DBdel(CF/anlage)
exten => 44,n,Playback(auth-thankyou)
exten => 44,n,NoOp(Weiterleitung fuer Anlage deaktiviert.)
exten => 44,n,Hangup()

[...]

[von_telekom]
exten => 063333333,1,NoOp(Eingehender Anruf ueber Telekom fuer ${EXTEN})
exten => 063333333,n,set(CALLERID(name)=0${CALLERID(num):3})
exten => 063333333,n,set(CALLERID(num)=0${CALLERID(num):3})
exten => 063333333,n,NoOp(DB(CF/anlage) = ${DB(CF/anlage)})
exten => 063333333,n,GotoIf($[foo${DB(CF/anlage)} != foo]?forward:normal)
exten => 063333333,n(forward),NoOp(Rufumleitung fuer Anlage aktiviert, es wird verbunden zu ${DB(CF/anlage)})
exten => 063333333,n,Goto(telefon,${DB(CF/anlage)},1)
exten => 063333333,n(normal),NoOp(Anruf fuer Anlage werden ganz normal intern zugestellt)
exten => 063333333,n,Dial(SIP/1000&SIP/1001&SIP/1005,60)
exten => 063333333,n,Hangup()

[...]

Also wie gesagt. Das Routing durch den Dialplan passt, am Ende klingelt die Nummer die zuvor gesetzt wurde. Allerdings ist schon beim Klingel keine Ton zu hören und der RTP Debug gibt gar nichts aus!
Code: 
[KTK-server*CLI> 
[0K    -- Executing [4401733333333@telefon:4] [1;36mNoOp[0m("[1;35mSIP/11-000003f6[0m", "[1;35mWeiterleitung fuer 063333333 auf 01733333333 aktiviert.[0m") in new stack
    -- Executing [4401733333333@telefon:5] [1;36mHangup[0m("[1;35mSIP/11-000003f6[0m", "[1;35m[0m") in new stack
  == Spawn extension (telefon, 4401733333333, 5) exited non-zero on 'SIP/11-000003f6'

[KTK-server*CLI> 
[0K  == Extension Changed 11[call_sip_accounts] new state Idle for Notify User 10 

[KTK-server*CLI> 
[0K  == Using SIP RTP CoS mark 5

[KTK-server*CLI> 
[0K    -- Executing [063333333@von_telekom:1] [1;36mNoOp[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35mEingehender Anruf ueber Telekom fuer 063333333[0m") in new stack
    -- Executing [063333333@von_telekom:2] [1;36mNoOp[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35m+496944444[0m") in new stack

[KTK-server*CLI> 
[0K    -- Executing [063333333@von_telekom:3] [1;36mSet[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35mCALLERID(name)=06944444[0m") in new stack

[KTK-server*CLI> 
[0K    -- Executing [063333333@von_telekom:4] [1;36mSet[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35mCALLERID(num)=06944444[0m") in new stack

[KTK-server*CLI> 
[0K    -- Executing [063333333@von_telekom:5] [1;36mNoOp[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35m'06944444 <06944444>'[0m") in new stack

[KTK-server*CLI> 
[0K    -- Executing [063333333@von_telekom:6] [1;36mNoOp[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35mDB(CF/anlage) = 01733333333[0m") in new stack

[KTK-server*CLI> 
[0K    -- Executing [063333333@von_telekom:7] [1;36mGotoIf[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35m1?forward:normal[0m") in new stack

[KTK-server*CLI> 
[0K    -- Goto (von_telekom,063333333,8)
    -- Executing [063333333@von_telekom:8] [1;36mNoOp[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35mRufumleitung fuer 063333333 aktiviert, es wird verbunden zu 01733333333[0m") in new stack

[KTK-server*CLI> 
[0K    -- Executing [063333333@von_telekom:9] [1;36mGoto[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35mtelefon,01733333333,1[0m") in new stack

[KTK-server*CLI> 
[0K    -- Goto (telefon,01733333333,1)

[KTK-server*CLI> 
[0K    -- Executing [01733333333@telefon:1] [1;36mGoto[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35m00491733333333,1[0m") in new stack
    -- Goto (telefon,00491733333333,1)
    -- Executing [00491733333333@telefon:1] [1;36mSet[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35mCALLERID(name)=4963333333[0m") in new stack

[KTK-server*CLI> 
[0K    -- Executing [00491733333333@telefon:2] [1;36mSet[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35mCALLERID(number)=000000[0m") in new stack
    -- Executing [00491733333333@telefon:3] [1;36mDial[0m("[1;35mSIP/telekom-000003f7[0m", "[1;35mSIP/00491733333333@telmy,60,trg[0m") in new stack

[KTK-server*CLI> 
[0K  == Using SIP RTP CoS mark 5

[KTK-server*CLI> 
[0K    -- Called SIP/00491733333333@telmy

[KTK-server*CLI> 
[0K    -- SIP/telmy-000003f8 answered SIP/telekom-000003f7

[KTK-server*CLI> 
[0K  == Spawn extension (telefon, 00491733333333, 3) exited non-zero on 'SIP/telekom-000003f7'

[KTK-server*CLI> exit

Den Tipp den ich sonst hier im Forum gefunden habe, den Parameter directmedia einzufügen, hat bisher auch nicht geholfen. Weiter Ideen/Vorschläge das Problem zu lösen?

Gruß Valentin
 
Zuletzt bearbeitet:
Probier mal zusätzlich directrtpsetup=no. Ansonsten stell mit sip set debug on das SIP Debugging an und poste das hier, da kann man sehen, wie der RTP stream ausgehandelt wird.
 
Okay, habe mal zusätzlich directrtpsetup=no im [general] Abschnitt eingefügt und den sip debug mitlaufen lassen. Ich hatte mir die Ausgabe auch schon mal zuvor angeschaut, werde aber nicht richtig schlau daraus:

Code: 
Asterisk 1.8.10.1~dfsg-1ubuntu1, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
[0;37m[0mConnected to Asterisk 1.8.10.1~dfsg-1ubuntu1 currently running on gall-server (pid = 4741)
gall-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK82fbce43f667b330abb356453b917f3f
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bK20ee27e75ad7139b5696587de0aeee80.874fbe19
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
Max-Forwards: 69
To: <sip:[email protected]:5060>
From: <sip:[email protected];user=phone>;tag=52f529d5
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
Supported: histinfo
CSeq: 7156283 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, PUBLISH, REGISTER, SUBSCRIBE, UPDATE
P-Asserted-Identity: <sip:[email protected];user=phone>
P-Called-Party-ID: <sip:[email protected]>
P-User-Database: <aaa://HNOTDSP01-DSP.tel.t-online.de>
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 188

v=0
o=- 1395213200 1395213200 IN IP4 217.0.16.106
s=Basic Session
c=IN IP4 217.0.0.140
t=0 0
m=audio 18630 RTP/AVP 8 99
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
<------------->
--- (18 headers 9 lines) ---

[KTK-server*CLI> 
[0KSending to 217.0.16.170:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer 'telekom' for '+496944444' from 217.0.16.170:5060

[KTK-server*CLI> 
[0KFound RTP audio format 8
Found RTP audio format 99
Found audio description format telephone-event for ID 99
Capabilities: us - 0x14e (gsm|ulaw|alaw|slin|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.0.0.140:18630

[KTK-server*CLI> 
[0KLooking for 063333333 in von_telekom (domain 10.2.2.2)

[KTK-server*CLI> 
[0Klist_route: hop: <sip:[email protected]:5060;lr>
list_route: hop: <sip:[email protected]:5060;transport=udp;lr>

[KTK-server*CLI> 
[0K
<--- Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK82fbce43f667b330abb356453b917f3f;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bK20ee27e75ad7139b5696587de0aeee80.874fbe19
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=52f529d5
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 7156283 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
Audio is at 10016
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x40 (slin) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 62.153.171.35:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK418debc3;rport
Max-Forwards: 70
From: "4963333333" <sip:[email protected]>;tag=as0940df41
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Wed, 19 Mar 2014 07:13:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1688402626 1688402626 IN IP4 10.2.2.2
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 10.2.2.2
t=0 0
m=audio 10016 RTP/AVP 8 0 3 10 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK82fbce43f667b330abb356453b917f3f;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bK20ee27e75ad7139b5696587de0aeee80.874fbe19
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=52f529d5
To: <sip:[email protected]:5060>;tag=as3aafb5e0
Call-ID: [email protected]
CSeq: 7156283 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:62.153.171.35:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK418debc3;received=87.146.103.20;rport=5060
From: "4963333333" <sip:[email protected]>;tag=as0940df41
To: <sip:[email protected]:5060>;tag=as537e3ae1
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="mediagw01", nonce="52899ab6"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to

[KTK-server*CLI> 
[0Kset_destination: set destination to 62.153.171.35:5060
Transmitting (NAT) to 62.153.171.35:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK418debc3;rport
Max-Forwards: 70
From: "4963333333" <sip:[email protected]>;tag=as0940df41
To: <sip:[email protected]:5060>;tag=as537e3ae1
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---

[KTK-server*CLI> 
[0KAudio is at 10016
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x40 (slin) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 62.153.171.35:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK21c98464;rport
Max-Forwards: 70
From: "4963333333" <sip:[email protected]>;tag=as0940df41
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Authorization: Digest username="sip-telmy.com-000000", realm="mediagw01", algorithm=MD5, uri="sip:[email protected]:5060", nonce="52899ab6", response="9aadbb1c259ecbd9a0602c6855615993"
Date: Wed, 19 Mar 2014 07:13:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1688402626 1688402627 IN IP4 10.2.2.2
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 10.2.2.2
t=0 0
m=audio 10016 RTP/AVP 8 0 3 10 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:62.153.171.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK21c98464;received=87.146.103.20;rport=5060
From: "4963333333" <sip:[email protected]>;tag=as0940df41
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:62.153.171.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK21c98464;received=87.146.103.20;rport=5060
From: "4963333333" <sip:[email protected]>;tag=as0940df41
To: <sip:[email protected]:5060>;tag=as6da7e31d
Call-ID: [email protected]
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 663433533 663433533 IN IP4 62.153.171.35
s=Asterisk PBX 1.8.5.0
c=IN IP4 62.153.171.35
t=0 0
m=audio 14968 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101

[KTK-server*CLI> 
[0KFound audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101

[KTK-server*CLI> 
[0KCapabilities: us - 0x14e (gsm|ulaw|alaw|slin|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[KTK-server*CLI> 
[0KPeer audio RTP is at port 62.153.171.35:14968
list_route: hop: <sip:[email protected]:5060>

[KTK-server*CLI> 
[0Kset_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 62.153.171.35:5060

[KTK-server*CLI> 
[0KTransmitting (NAT) to 62.153.171.35:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK2cdc18b9;rport
Max-Forwards: 70
From: "4963333333" <sip:[email protected]>;tag=as0940df41
To: <sip:[email protected]:5060>;tag=as6da7e31d
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---

[KTK-server*CLI> 
[0KAudio is at 15170
Adding codec 0x8 (alaw) to SDP

[KTK-server*CLI> 
[0KAdding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK82fbce43f667b330abb356453b917f3f;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bK20ee27e75ad7139b5696587de0aeee80.874fbe19
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=52f529d5
To: <sip:[email protected]:5060>;tag=as3aafb5e0
Call-ID: [email protected]
CSeq: 7156283 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 1298757102 1298757102 IN IP4 10.2.2.2
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 10.2.2.2
t=0 0
m=audio 15170 RTP/AVP 8 99
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-16
a=ptime:20
a=sendrecv

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK5fac221be9a0b6d452093e0b421809d5
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bKaf575eb48809e472d0684fcb88cbddfd.79d3831a
Route: <sip:[email protected]:5060;transport=udp;lr>, <sip:[email protected]:5060;lr>, <sip:87.146.103.20:5060;transport=udp;lr>;nexthop
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
Max-Forwards: 69
To: <sip:[email protected]:5060>;tag=as3aafb5e0
From: <sip:[email protected];user=phone>;tag=52f529d5
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 7156283 ACK
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, PUBLISH, REGISTER, SUBSCRIBE, UPDATE
Content-Length: 0

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.10.79.9:5060 --->

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKd6b343b390a581346181bcc5a868870f
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bKbeecf3725403b6b18c142c1ae3b01544.13f1da70
Route: <sip:[email protected]:5060;transport=udp;lr>, <sip:[email protected]:5060;lr>, <sip:87.146.103.20:5060;transport=udp;lr>;nexthop
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
Max-Forwards: 69
To: <sip:[email protected]:5060>;tag=as3aafb5e0
From: <sip:[email protected];user=phone>;tag=52f529d5
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 7156284 INVITE
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 188

v=0
o=- 1395213200 1395213200 IN IP4 217.0.16.106
s=Basic Session
c=IN IP4 217.0.0.140
t=0 0
m=audio 18630 RTP/AVP 8 99
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
<------------->
--- (14 headers 9 lines) ---
Sending to 217.0.16.170:5060 (NAT)

[KTK-server*CLI> 
[0K
<--- Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKd6b343b390a581346181bcc5a868870f;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bKbeecf3725403b6b18c142c1ae3b01544.13f1da70
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=52f529d5
To: <sip:[email protected]:5060>;tag=as3aafb5e0
Call-ID: [email protected]
CSeq: 7156284 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


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

[KTK-server*CLI> 
[0KAudio is at 15170
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[KTK-server*CLI> 
[0K
<--- Reliably Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKd6b343b390a581346181bcc5a868870f;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bKbeecf3725403b6b18c142c1ae3b01544.13f1da70
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=52f529d5
To: <sip:[email protected]:5060>;tag=as3aafb5e0
Call-ID: [email protected]
CSeq: 7156284 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 1298757102 1298757102 IN IP4 10.2.2.2
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 10.2.2.2
t=0 0
m=audio 15170 RTP/AVP 8 99
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-16
a=ptime:20
a=sendrecv

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKd6b343b390a581346181bcc5a868870f
Via: SIP/2.0/TCP 62.155.0.194:5060;branch=z9hG4bKbeecf3725403b6b18c142c1ae3b01544.13f1da70
Route: <sip:[email protected]:5060;transport=udp;lr>, <sip:[email protected]:5060;lr>, <sip:87.146.103.20:5060;transport=udp;lr>;nexthop
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
To: <sip:[email protected]:5060>;tag=as3aafb5e0
From: <sip:[email protected];user=phone>;tag=52f529d5
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 7156284 ACK
Content-Length: 0

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

[KTK-server*CLI> 
[0KReliably Transmitting (NAT) to 10.2.2.10:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK3c982b24;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as7d99ab5a
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Wed, 19 Mar 2014 07:13:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[KTK-server*CLI> 
[0KReliably Transmitting (NAT) to 62.153.171.35:5060:
OPTIONS sip:sip.telmy.com SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK6a32bc5e;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as0bdf1d18
To: <sip:sip.telmy.com>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Wed, 19 Mar 2014 07:13:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:10.2.2.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK3c982b24;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as7d99ab5a
To: <sip:[email protected]:5060>;tag=387518783
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP2100 1.0.5.32
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:62.153.171.35:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK6a32bc5e;rport;received=87.146.103.20
From: "asterisk" <sip:[email protected]>;tag=as0bdf1d18
To: <sip:sip.telmy.com>;tag=as416c1e5a
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

[KTK-server*CLI> 
[0KReliably Transmitting (NAT) to 10.2.2.11:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK71d50db0;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as12589012
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Wed, 19 Mar 2014 07:13:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[KTK-server*CLI> 
[0KReliably Transmitting (NAT) to 10.2.2.19:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK04796d81;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as619d6d99
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Wed, 19 Mar 2014 07:13:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:10.2.2.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK71d50db0;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as12589012
To: <sip:[email protected]:5060>;tag=1293693124
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP2100 1.0.5.32
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:10.2.2.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK04796d81;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as619d6d99
To: <sip:[email protected]:5060>;tag=709332058
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-502 V1.1B 1.0.10.9 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

[KTK-server*CLI> 
[0KReliably Transmitting (NAT) to 217.0.16.170:5060:
OPTIONS sip:tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK153b1441;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as25717023
To: <sip:tel.t-online.de>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Wed, 19 Mar 2014 07:13:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;rport=5060;received=87.146.103.20;branch=z9hG4bK153b1441
To: <sip:tel.t-online.de>;tag=f2774363
From: asterisk <sip:[email protected]>;tag=as25717023
Call-ID: [email protected]:5060
Supported: 100rel,path
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, PUBLISH, REFER, REGISTER, SUBSCRIBE, UPDATE
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

[KTK-server*CLI> exit
[0K
<--- SIP read from UDP:62.153.171.35:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 62.153.171.35:5060;branch=z9hG4bK7d6b930f;rport
Max-Forwards: 70
From: <sip:[email protected]:5060>;tag=as6da7e31d
To: "4963333333" <sip:[email protected]>;tag=as0940df41
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.5.0
Proxy-Authorization: Digest username="000000", realm="mediagw01", algorithm=MD5, uri="sip.telmy.com:5060", nonce="", response="ff66cc14cd79244c1c06a535dc62f38b"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 62.153.171.35:5060 (NAT)

[KTK-server*CLI> exit
[0KScheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 62.153.171.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.153.171.35:5060;branch=z9hG4bK7d6b930f;received=62.153.171.35;rport=5060
From: <sip:[email protected]:5060>;tag=as6da7e31d
To: "4963333333" <sip:[email protected]>;tag=as0940df41
Call-ID: [email protected]
CSeq: 102 BYE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

[KTK-server*CLI> exit
[0K
<--- SIP read from UDP:217.10.79.9:5060 --->

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

[KTK-server*CLI> exit
[0KScheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:[email protected]:5060;lr> for address/port to send to

[KTK-server*CLI> exit
[0Kset_destination: set destination to 217.0.16.170:5060

[KTK-server*CLI> exit
[0KReliably Transmitting (NAT) to 217.0.16.170:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK50d09d5f;rport
Route: <sip:[email protected]:5060;lr>,<sip:[email protected]:5060;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]:5060>;tag=as3aafb5e0
To: <sip:[email protected];user=phone>;tag=52f529d5
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

[KTK-server*CLI> exit
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;rport=5060;received=87.146.103.20;branch=z9hG4bK50d09d5f
Record-Route: <sip:[email protected]:5060;transport=udp;lr>, <sip:[email protected]:5060;lr>
To: <sip:[email protected];user=phone>;tag=52f529d5
From: <sip:[email protected]:5060>;tag=as3aafb5e0
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 102 BYE
Content-Length: 0
 
Da Du Dich scheinbar hinter einer NAT befindest (10.2.2.2), brauchst Du wohl noch externip oder externhost/externrefresh.
 
Naja 10.2.2.2 ist mein Asterisk Server und 10.2.2.1 mein Router. Ich werde mal mit externhost testen und dann berichten.

// EDIT //

Die Attribute bringen keine Verbesserung! Ich hab in der Log jetzt mal von Handy (01733333333) angerufen und der Anruf wurde auf Festnetz (06944444) weitergeleitet.

Code: 
Asterisk 1.8.10.1~dfsg-1ubuntu1, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
[0;37m[0mConnected to Asterisk 1.8.10.1~dfsg-1ubuntu1 currently running on gall-server (pid = 4741)
gall-server*CLI> 
[0KVerbosity is at least 12

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:149.3.130.130:5153 --->
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 149.3.130.130:5153;branch=z9hG4bK-4226104404;rport
Content-Length: 0
From: "sipvicious"<sip:[email protected]>; tag=3537393236373164313363340134323934363833323032
Accept: application/sdp
User-Agent: friendly-scanner
To: "sipvicious"<sip:[email protected]>
Contact: sip:[email protected]:5153
CSeq: 1 OPTIONS
Call-ID: 675019260109964287863529
Max-Forwards: 70

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.10.79.9:5060 --->

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK3ac978007e0450e95d5f321c6faf2ff5
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bK976826f1dbe528ecfb1b208453d68084.16536ba1
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
Max-Forwards: 69
To: <sip:[email protected]:5060>
From: <sip:[email protected];user=phone>;tag=fafeeb71
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
Supported: histinfo
CSeq: 5617071 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, PUBLISH, REGISTER, SUBSCRIBE, UPDATE
P-Asserted-Identity: <sip:[email protected];user=phone>
P-Called-Party-ID: <sip:[email protected]>
P-User-Database: <aaa://LEITDSP01-DSP.tel.t-online.de>
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 188

v=0
o=- 1395320108 1395320108 IN IP4 217.0.17.230
s=Basic Session
c=IN IP4 217.0.1.135
t=0 0
m=audio 15800 RTP/AVP 8 99
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
<------------->
--- (18 headers 9 lines) ---

[KTK-server*CLI> 
[0KSending to 217.0.16.170:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer 'telekom' for '+491733333333' from 217.0.16.170:5060

[KTK-server*CLI> 
[0K  == Using SIP RTP CoS mark 5

[KTK-server*CLI> 
[0KFound RTP audio format 8
Found RTP audio format 99
Found audio description format telephone-event for ID 99
Capabilities: us - 0x14e (gsm|ulaw|alaw|slin|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)

[KTK-server*CLI> 
[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.0.1.135:15800
Looking for 063333333 in von_telekom (domain 10.2.2.2)

[KTK-server*CLI> 
[0Klist_route: hop: <sip:[email protected]:5060;lr>
list_route: hop: <sip:[email protected]:5060;transport=udp;lr>

[KTK-server*CLI> 
[0K
<--- Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK3ac978007e0450e95d5f321c6faf2ff5;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bK976826f1dbe528ecfb1b208453d68084.16536ba1
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=fafeeb71
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 5617071 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
    -- Executing [063333333@von_telekom:1] [1;36mNoOp[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mEingehender Anruf ueber Telekom fuer 063333333[0m") in new stack
    -- Executing [063333333@von_telekom:2] [1;36mNoOp[0m("[1;35mSIP/telekom-00000146[0m", "[1;35m+491733333333[0m") in new stack
    -- Executing [063333333@von_telekom:3] [1;36mSet[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mCALLERID(name)=01733333333[0m") in new stack
    -- Executing [063333333@von_telekom:4] [1;36mSet[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mCALLERID(num)=01733333333[0m") in new stack
    -- Executing [063333333@von_telekom:5] [1;36mNoOp[0m("[1;35mSIP/telekom-00000146[0m", "[1;35m'01733333333 <01733333333>'[0m") in new stack
    -- Executing [063333333@von_telekom:6] [1;36mNoOp[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mDB(CF/anlage) = 06944444[0m") in new stack

[KTK-server*CLI> 
[0K    -- Executing [063333333@von_telekom:7] [1;36mGotoIf[0m("[1;35mSIP/telekom-00000146[0m", "[1;35m1?forward:normal[0m") in new stack
    -- Goto (von_telekom,063333333,8)
    -- Executing [063333333@von_telekom:8] [1;36mNoOp[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mRufumleitung fuer 063333333 aktiviert, es wird verbunden zu 06944444[0m") in new stack

[KTK-server*CLI> 
[0K    -- Executing [063333333@von_telekom:9] [1;36mGoto[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mtelefon,06944444,1[0m") in new stack
    -- Goto (telefon,06944444,1)
    -- Executing [06944444@telefon:1] [1;36mGoto[0m("[1;35mSIP/telekom-00000146[0m", "[1;35m00496944444,1[0m") in new stack
    -- Goto (telefon,00496944444,1)
    -- Executing [00496944444@telefon:1] [1;36mNoOp[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mAbgehende Verbindung (festnetz) via Telekom 01733333333[0m") in new stack
    -- Executing [00496944444@telefon:2] [1;36mSet[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mCALLERID(name)=063333333[0m") in new stack
    -- Executing [00496944444@telefon:3] [1;36mSet[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mCALLERID(num)=063333333[0m") in new stack
    -- Executing [00496944444@telefon:4] [1;36mDial[0m("[1;35mSIP/telekom-00000146[0m", "[1;35mSIP/00496944444@telekom,60,trg[0m") in new stack

[KTK-server*CLI> 
[0K  == Using SIP RTP CoS mark 5

[KTK-server*CLI> 
[0KAudio is at 12982
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x40 (slin) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[KTK-server*CLI> 
[0KReliably Transmitting (NAT) to 217.0.16.170:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK0c80fbcd;rport
Max-Forwards: 70
From: "063333333" <sip:[email protected]>;tag=as3f8e94c2
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Thu, 20 Mar 2014 12:55:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1988983375 1988983375 IN IP4 10.2.2.2
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 10.2.2.2
t=0 0
m=audio 12982 RTP/AVP 8 0 3 10 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[KTK-server*CLI> 
[0K    -- Called SIP/00496944444@telekom

[KTK-server*CLI> 
[0K
<--- Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK3ac978007e0450e95d5f321c6faf2ff5;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bK976826f1dbe528ecfb1b208453d68084.16536ba1
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=fafeeb71
To: <sip:[email protected]:5060>;tag=as1b836d3f
Call-ID: [email protected]
CSeq: 5617071 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.2.2:5060;rport=5060;received=87.146.103.20;branch=z9hG4bK0c80fbcd
To: <sip:[email protected]:5060>;tag=b72d4cad
From: 063333333 <sip:[email protected]>;tag=as3f8e94c2
Call-ID: [email protected]
CSeq: 102 INVITE
WWW-Authenticate: Digest algorithm=MD5, nonce="07c5222507c5222554efc70987e7840fddf5d2d3687b79bf2d85d47676913cc7", realm="tel.t-online.de"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to

[KTK-server*CLI> 
[0Kset_destination: set destination to 217.0.16.170:5060
Transmitting (NAT) to 217.0.16.170:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK0c80fbcd;rport
Max-Forwards: 70
From: "063333333" <sip:[email protected]>;tag=as3f8e94c2
To: <sip:[email protected]:5060>;tag=b72d4cad
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---

[KTK-server*CLI> 
[0KAudio is at 12982
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x40 (slin) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[KTK-server*CLI> 
[0KReliably Transmitting (NAT) to 217.0.16.170:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK677a29ba;rport
Max-Forwards: 70
From: "063333333" <sip:[email protected]>;tag=as3f8e94c2
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Authorization: Digest username="username", realm="tel.t-online.de", algorithm=MD5, uri="sip:[email protected]:5060", nonce="07c5222507c5222554efc70987e7840fddf5d2d3687b79bf2d85d47676913cc7", response="6b236599694c2989e5fad81ac76602c0"
Date: Thu, 20 Mar 2014 12:55:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 1988983375 1988983376 IN IP4 10.2.2.2
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 10.2.2.2
t=0 0
m=audio 12982 RTP/AVP 8 0 3 10 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
SIP/2.0 100 Rufaufbau
Via: SIP/2.0/UDP 10.2.2.2:5060;rport=5060;received=87.146.103.20;branch=z9hG4bK677a29ba
To: <sip:[email protected]:5060>
From: 063333333 <sip:[email protected]>;tag=as3f8e94c2
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0

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

[KTK-server*CLI> 
[0K[Mar 20 13:55:13] [1;33mNOTICE[0m[4810]: [1;37mchan_sip.c[0m:[1;37m13058[0m [1;37msip_reregister[0m:    -- Re-registration for  [email protected]
       > doing dnsmgr_lookup for 'sip.telmy.com'

[KTK-server*CLI> 
[0KREGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 62.153.171.34:5060:
REGISTER sip:sip.telmy.com SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK741e0e7a;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as4bbef3a1
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Authorization: Digest username="sip-telmy.com-000000", realm="mediagw02", algorithm=MD5, uri="sip:sip.telmy.com", nonce="5049c579", response="75ff725ab3e496528505577f0797d0c5"
Expires: 240
Contact: <sip:[email protected]:5060>
Content-Length: 0


---

[KTK-server*CLI> 
[0K[Mar 20 13:55:13] [1;33mNOTICE[0m[4810]: [1;37mchan_sip.c[0m:[1;37m13058[0m [1;37msip_reregister[0m:    -- Re-registration for  [email protected]
       > doing dnsmgr_lookup for 'sipgate.de'

[KTK-server*CLI> 
[0K       > ast_get_srv: SRV lookup for '_sip._udp.sipgate.de' mapped to host sipgate.de, port 5060

[KTK-server*CLI> 
[0KREGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 217.10.79.9:5060:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK5b9af8a5;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as2a35ae19
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Authorization: Digest username="2222222", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="532ae6a8f9f7b9e61395b49ad3ffb4c2673163b9", response="edef2b08161924a60ef2ddd6c4585494"
Expires: 240
Contact: <sip:[email protected]:5060>
Content-Length: 0


---

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;received=87.146.103.20;branch=z9hG4bK5b9af8a5;rport=5060
From: <sip:[email protected]>;tag=as2a35ae19
To: <sip:[email protected]>;tag=c3e497ecaece77a8e244e564b4212178.3e89
Call-ID: [email protected]
CSeq: 104 REGISTER
Contact: <sip:[email protected]:5060>;expires=240;received="sip:87.146.103.20:5060"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[Mar 20 13:55:13] [1;33mNOTICE[0m[4810]: [1;37mchan_sip.c[0m:[1;37m20714[0m [1;37mhandle_response_register[0m: Outbound Registration: Expiry for sipgate.de is 240 sec (Scheduling reregistration in 225 s)

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:62.153.171.34:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK741e0e7a;received=87.146.103.20;rport=5060
From: <sip:[email protected]>;tag=as4bbef3a1
To: <sip:[email protected]>;tag=as27af1be8
Call-ID: [email protected]
CSeq: 104 REGISTER
Server: Asterisk PBX 1.8.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="mediagw02", nonce="101d7a43"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sip.telmy.com
       > doing dnsmgr_lookup for 'sip.telmy.com'

[KTK-server*CLI> 
[0KREGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 62.153.171.34:5060:
REGISTER sip:sip.telmy.com SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK3213ccf1;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as44631b40
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 105 REGISTER
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Authorization: Digest username="sip-telmy.com-000000", realm="mediagw02", algorithm=MD5, uri="sip:sip.telmy.com", nonce="101d7a43", response="6438a2ca58e7103fa48d3e3f8b5d4dbb"
Expires: 240
Contact: <sip:[email protected]:5060>
Content-Length: 0


---

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:62.153.171.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK3213ccf1;received=87.146.103.20;rport=5060
From: <sip:[email protected]>;tag=as44631b40
To: <sip:[email protected]>;tag=as27af1be8
Call-ID: [email protected]
CSeq: 105 REGISTER
Server: Asterisk PBX 1.8.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 240
Contact: <sip:[email protected]:5060>;expires=240
Date: Thu, 20 Mar 2014 12:55:13 GMT
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[Mar 20 13:55:13] [1;33mNOTICE[0m[4810]: [1;37mchan_sip.c[0m:[1;37m20714[0m [1;37mhandle_response_register[0m: Outbound Registration: Expiry for sip.telmy.com is 240 sec (Scheduling reregistration in 225 s)

[KTK-server*CLI> 
[0K[Mar 20 13:55:13] [1;33mNOTICE[0m[4810]: [1;37mchan_sip.c[0m:[1;37m13058[0m [1;37msip_reregister[0m:    -- Re-registration for  [email protected]
       > doing dnsmgr_lookup for 'tel.t-online.de'

[KTK-server*CLI> 
[0K       > ast_get_srv: SRV lookup for '_sip._udp.tel.t-online.de' mapped to host f-ipp-a01.isp.t-ipnet.de, port 5060

[KTK-server*CLI> 
[0KREGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 217.0.16.170:5060:
REGISTER sip:tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK02d2b860;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as1a86b725
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Authorization: Digest username="username", realm="tel.t-online.de", algorithm=MD5, uri="sip:tel.t-online.de", nonce="97ecf81397ecf813c4c61c4317f389ceba39cd98ca55113f5d08b4ba6c902c09", response="24f688a2ecd785c8c867d19637a908a5"
Expires: 240
Contact: <sip:[email protected]:5060>
Content-Length: 0


---

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;rport=5060;received=87.146.103.20;branch=z9hG4bK02d2b860
To: <sip:[email protected]>;tag=29d35aa8
From: <sip:[email protected]>;tag=as1a86b725
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>;expires=240
CSeq: 104 REGISTER
P-Associated-URI: <sip:[email protected];user=phone>
Content-Length: 0

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

[KTK-server*CLI> 
[0KScheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[Mar 20 13:55:13] [1;33mNOTICE[0m[4810]: [1;37mchan_sip.c[0m:[1;37m20714[0m [1;37mhandle_response_register[0m: Outbound Registration: Expiry for tel.t-online.de is 240 sec (Scheduling reregistration in 225 s)

[KTK-server*CLI> 
[0K[Mar 20 13:55:13] [1;33mNOTICE[0m[4810]: [1;37mchan_sip.c[0m:[1;37m13058[0m [1;37msip_reregister[0m:    -- Re-registration for  [email protected]
       > doing dnsmgr_lookup for 'tel.t-online.de'

[KTK-server*CLI> 
[0K       > ast_get_srv: SRV lookup for '_sip._udp.tel.t-online.de' mapped to host f-ipp-a01.isp.t-ipnet.de, port 5060

[KTK-server*CLI> 
[0KREGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 217.0.16.170:5060:
REGISTER sip:tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK572ea73d;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as51f9fbae
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Authorization: Digest username="username", realm="tel.t-online.de", algorithm=MD5, uri="sip:tel.t-online.de", nonce="9d59a6089d59a608ce7342581d46d43b4e9546c62a39ae8818237513592d76e3", response="501dbc2d68acbb4e1ad048bd4cdf7164"
Expires: 240
Contact: <sip:[email protected]:5060>
Content-Length: 0


---

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;rport=5060;received=87.146.103.20;branch=z9hG4bK572ea73d
To: <sip:[email protected]>;tag=531c1f14
From: <sip:[email protected]>;tag=as51f9fbae
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>;expires=240
CSeq: 104 REGISTER
P-Associated-URI: <sip:[email protected];user=phone>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[Mar 20 13:55:13] [1;33mNOTICE[0m[4810]: [1;37mchan_sip.c[0m:[1;37m20714[0m [1;37mhandle_response_register[0m: Outbound Registration: Expiry for tel.t-online.de is 240 sec (Scheduling reregistration in 225 s)

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
SIP/2.0 183 Verbindung wird aufgebaut (0)
Via: SIP/2.0/UDP 10.2.2.2:5060;rport=5060;received=87.146.103.20;branch=z9hG4bK677a29ba
To: <sip:[email protected]:5060>;tag=4199a2ef
From: 063333333 <sip:[email protected]>;tag=as3f8e94c2
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 103 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REGISTER, UPDATE
Session-ID: [email protected]
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 276

v=0
o=- 5283320140220135508 1836974186 IN IP4 217.0.16.170
s=Phone Call via hiQ9200 SIPCA
c=IN IP4 217.0.1.211
t=0 0
m=audio 15772 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=sendrecv
a=ptime:20
<------------->
--- (12 headers 13 lines) ---
list_route: hop: <sip:[email protected]:5060>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101

[KTK-server*CLI> 
[0KCapabilities: us - 0x14e (gsm|ulaw|alaw|slin|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[KTK-server*CLI> 
[0KPeer audio RTP is at port 217.0.1.211:15772

[KTK-server*CLI> 
[0K    -- SIP/telekom-00000147 is making progress passing it to SIP/telekom-00000146

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
SIP/2.0 180 Klingeln
Via: SIP/2.0/UDP 10.2.2.2:5060;rport=5060;received=87.146.103.20;branch=z9hG4bK677a29ba
To: <sip:[email protected]:5060>;tag=4199a2ef
From: 063333333 <sip:[email protected]>;tag=as3f8e94c2
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 103 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REGISTER, UPDATE
Session-ID: [email protected]
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 276

v=0
o=- 5283320140220135508 1836974186 IN IP4 217.0.16.170
s=Phone Call via hiQ9200 SIPCA
c=IN IP4 217.0.1.211
t=0 0
m=audio 15772 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=sendrecv
a=ptime:20
<------------->
--- (12 headers 13 lines) ---
list_route: hop: <sip:[email protected]:5060>
    -- SIP/telekom-00000147 is ringing

[KTK-server*CLI> 
[0K
<--- Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK3ac978007e0450e95d5f321c6faf2ff5;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bK976826f1dbe528ecfb1b208453d68084.16536ba1
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=fafeeb71
To: <sip:[email protected]:5060>;tag=as1b836d3f
Call-ID: [email protected]
CSeq: 5617071 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


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

[KTK-server*CLI> 
[0K    -- SIP/telekom-00000147 is making progress passing it to SIP/telekom-00000146

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.2:5060;rport=5060;received=87.146.103.20;branch=z9hG4bK677a29ba
To: <sip:[email protected]:5060>;tag=4199a2ef
From: 063333333 <sip:[email protected]>;tag=as3f8e94c2
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 103 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REGISTER, UPDATE
Session-ID: [email protected]
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 276

v=0
o=- 5283320140220135508 1836974186 IN IP4 217.0.16.170
s=Phone Call via hiQ9200 SIPCA
c=IN IP4 217.0.1.211
t=0 0
m=audio 15772 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=sendrecv
a=ptime:20
<------------->
--- (12 headers 13 lines) ---
list_route: hop: <sip:[email protected]:5060>
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 217.0.16.170:5060

[KTK-server*CLI> 
[0KTransmitting (NAT) to 217.0.16.170:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.2.2:5060;branch=z9hG4bK6f708931;rport
Max-Forwards: 70
From: "063333333" <sip:[email protected]>;tag=as3f8e94c2
To: <sip:[email protected]:5060>;tag=4199a2ef
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---

[KTK-server*CLI> 
[0K    -- SIP/telekom-00000147 answered SIP/telekom-00000146

[KTK-server*CLI> 
[0KAudio is at 15430
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[KTK-server*CLI> 
[0K
<--- Reliably Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bK3ac978007e0450e95d5f321c6faf2ff5;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bK976826f1dbe528ecfb1b208453d68084.16536ba1
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=fafeeb71
To: <sip:[email protected]:5060>;tag=as1b836d3f
Call-ID: [email protected]
CSeq: 5617071 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 234761313 234761313 IN IP4 10.2.2.2
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 10.2.2.2
t=0 0
m=audio 15430 RTP/AVP 8 99
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-16
a=ptime:20
a=sendrecv

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKa2be67de840b9b54ceccc98c6708045b
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bKbbeda50cbb99ebee92251f5e09143df9.4459712b
Route: <sip:[email protected]:5060;transport=udp;lr>, <sip:[email protected]:5060;lr>, <sip:87.146.103.20:5060;transport=udp;lr>;nexthop
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
Max-Forwards: 69
To: <sip:[email protected]:5060>;tag=as1b836d3f
From: <sip:[email protected];user=phone>;tag=fafeeb71
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 5617071 ACK
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, PUBLISH, REGISTER, SUBSCRIBE, UPDATE
Content-Length: 0

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.10.79.9:5060 --->

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKf3274a278b834efa7ab7fd2164cd251d.b28d670d
Max-Forwards: 67
To: 063333333 <sip:[email protected]>;tag=as3f8e94c2
From: <sip:[email protected]:5060>;tag=4199a2ef
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 10271513 INVITE
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 282

v=0
o=hiQ9200 5283320140220135508 1836974186 IN IP4 217.0.16.170
s=Phone Call via hiQ9200 SIPCA
c=IN IP4 217.0.1.211
t=0 0
m=audio 15772 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=sendrecv
a=ptime:20
<------------->
--- (11 headers 13 lines) ---
Sending to 217.0.16.170:5060 (NAT)

[KTK-server*CLI> 
[0K
<--- Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKf3274a278b834efa7ab7fd2164cd251d.b28d670d;received=217.0.16.170;rport=5060
From: <sip:[email protected]:5060>;tag=4199a2ef
To: 063333333 <sip:[email protected]>;tag=as3f8e94c2
Call-ID: [email protected]
CSeq: 10271513 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


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

[KTK-server*CLI> 
[0KAudio is at 12982
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[KTK-server*CLI> 
[0K
<--- Reliably Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKf3274a278b834efa7ab7fd2164cd251d.b28d670d;received=217.0.16.170;rport=5060
From: <sip:[email protected]:5060>;tag=4199a2ef
To: 063333333 <sip:[email protected]>;tag=as3f8e94c2
Call-ID: [email protected]
CSeq: 10271513 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 1988983375 1988983376 IN IP4 10.2.2.2
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 10.2.2.2
t=0 0
m=audio 12982 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKf3274a278b834efa7ab7fd2164cd251d.b28d670d
To: 063333333 <sip:[email protected]>;tag=as3f8e94c2
From: <sip:[email protected]:5060>;tag=4199a2ef
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 10271513 ACK
Content-Length: 0

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKda583027c7b39418bd668ef699faf89e
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bKf6e04aa3d49b68e57ddba38b8a826ff0.6a7dcd8f
Route: <sip:[email protected]:5060;transport=udp;lr>, <sip:[email protected]:5060;lr>, <sip:87.146.103.20:5060;transport=udp;lr>;nexthop
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
Max-Forwards: 69
To: <sip:[email protected]:5060>;tag=as1b836d3f
From: <sip:[email protected];user=phone>;tag=fafeeb71
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 5617072 INVITE
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 188

v=0
o=- 1395320108 1395320108 IN IP4 217.0.17.230
s=Basic Session
c=IN IP4 217.0.1.135
t=0 0
m=audio 15800 RTP/AVP 8 99
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
<------------->
--- (14 headers 9 lines) ---
Sending to 217.0.16.170:5060 (NAT)

[KTK-server*CLI> 
[0K
<--- Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKda583027c7b39418bd668ef699faf89e;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bKf6e04aa3d49b68e57ddba38b8a826ff0.6a7dcd8f
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=fafeeb71
To: <sip:[email protected]:5060>;tag=as1b836d3f
Call-ID: [email protected]
CSeq: 5617072 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


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

[KTK-server*CLI> e
[0KAudio is at 15430
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

[KTK-server*CLI> 
[0K
<--- Reliably Transmitting (NAT) to 217.0.16.170:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKda583027c7b39418bd668ef699faf89e;received=217.0.16.170;rport=5060
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bKf6e04aa3d49b68e57ddba38b8a826ff0.6a7dcd8f
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=fafeeb71
To: <sip:[email protected]:5060>;tag=as1b836d3f
Call-ID: [email protected]
CSeq: 5617072 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 234761313 234761313 IN IP4 10.2.2.2
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 10.2.2.2
t=0 0
m=audio 15430 RTP/AVP 8 99
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-16
a=ptime:20
a=sendrecv

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

[KTK-server*CLI> 
[0K
<--- SIP read from UDP:217.0.16.170:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.0.16.170:5060;branch=z9hG4bKda583027c7b39418bd668ef699faf89e
Via: SIP/2.0/TCP 62.155.3.194:5060;branch=z9hG4bKf6e04aa3d49b68e57ddba38b8a826ff0.6a7dcd8f
Route: <sip:[email protected]:5060;transport=udp;lr>, <sip:[email protected]:5060;lr>, <sip:87.146.103.20:5060;transport=udp;lr>;nexthop
Record-Route: <sip:[email protected]:5060;lr>, <sip:[email protected]:5060;transport=udp;lr>
To: <sip:[email protected]:5060>;tag=as1b836d3f
From: <sip:[email protected];user=phone>;tag=fafeeb71
Call-ID: [email protected]
Contact: <sip:[email protected]:5060>
CSeq: 5617072 ACK
Content-Length: 0

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

[KTK-server*CLI> exit
 
Zuletzt bearbeitet:
Hab gerade noch eine Idee bekommen und mal meine extension.conf so modifiziert das der Anruf erst mit Answer() angenommen wird und dann weitergeleitet wird.

Soweit scheint es zu funktionieren!

Code: 
[von_telekom]
exten => 063333333,1,NoOp(Eingehender Anruf ueber Telekom fuer ${EXTEN})
exten => 063333333,n,set(CALLERID(name)=0${CALLERID(num):3})
exten => 063333333,n,set(CALLERID(num)=0${CALLERID(num):3})
exten => 063333333,n,NoOp(DB(CF/anlage) = ${DB(CF/anlage)})
exten => 063333333,n,GotoIf($[foo${DB(CF/anlage)} != foo]?forward:normal)
exten => 063333333,n(forward),NoOp(Rufumleitung fuer Anlage aktiviert, es wird verbunden zu ${DB(CF/anlage)})
exten => 063333333,n,Answer()
exten => 063333333,n,Goto(telefon,${DB(CF/anlage)},1)
exten => 063333333,n(normal),NoOp(Anruf fuer Anlage werden ganz normal intern zugestellt)
exten => 063333333,n,Dial(SIP/1000&SIP/1001&SIP/1005,60)
exten => 063333333,n,Hangup()

PS: Das scheint auch ohne externip und externhost zu gehen. Trotzdem Danke für den Support rentier-s
 
Hm, da bin ich zu wenig Expertin um das jetzt zu verstehen, aber so lange es funktioniert.

Ändere das Titel Prefix dann doch bitte in "gelöst" (im 1. Beitrag auf Bearbeiten -> Erweitert).
 
rentier-s schrieb:
Hm, da bin ich zu wenig Expertin um das jetzt zu verstehen, aber so lange es funktioniert.
Zum Vergrößern anklicken....

Ich war darauf gekommen, weil ich irgendwo hier im Forum mal gelesen habe, dass nur wenn der Anruf angenommen wird der Asterisk auch die RTP Ports verhandelt. Das würde auch erklären warum ohne das antworten keine RTP Pakete protokolliert werden.

Auf jeden Fall noch mal vielen Danke für die mentale und technische Unterstützung. ;)
 
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