[DX800] Anrufweiterschaltung via Asterisk (13.1.0) mit DX800A tut nicht

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tazacorte

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Hallo zusammen.

wir haben seit einigen Monaten einen SIP-Trunk-Anschluß bei equada. Als "Zentrale" dient ein Gigaset DX800A. Telefonieren funktioniert tadellos. Auch die Weiterleitung eines externen Anrufs auf eine interne Nebenstelle oder einen externen Teilnehmer funktioniert, jedoch nur, wenn man den Anruf mit dem Gigaset entgegennimmt und per Rückfrage weiterleitet. Wenn ich die AWS im DX800A aktiviere (Einstellung "immer") und eine externe Rufnummer hinterlege, scheitert dies mit dem Hinweis im Asterisk Protokoll:
Not accepting call completion offers from call-forward recipient

Der Equada-Support schreibt:
"wenn Ihre Telefonanlage eine Funktion Ihres Endgerätes nicht unterstützt. Dann müssen Sie sich bitte an den Entwickler der Anlage oder des Telefons wenden. Der SIP-Trunk hat hier keinen Einfluss."
Zum Vergrößern anklicken....


Nach Recherche in VoiP-Foren habe ich in der sip.conf folgende Werte von "no" auf "yes" geändert:

promiscredir
progressinband
sendrpid
trustrpid

Leider ohne Erfolg. Wenn "promiscredir" auf "yes" gesetzt wird die Verbindung sogar sofort abgebrochen.
Ansonsten klingelt es fleißig vor sich hin, aber eine Verbindung zu der externen Nummer kommt nicht zustande.

Ich freue mich auf Eure Beiträge!

Hier das ganze Protokoll des Anrufs:
Code: 
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:46.182.249.38:5083 --->
ACK sip:0714****@192.168.178.40:5060 SIP/2.0
Via: SIP/2.0/UDP 46.182.249.38:5083;branch=z9hG4bKmrvo9330a0k0ang36790.1
From: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
To: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 2 ACK
Max-Forwards: 69
Contact: <sip:0714****[email protected]:5083;transport=udp>
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Retransmitting #1 (NAT) to 192.168.178.39:5062:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK049d19e8;rport
Max-Forwards: 70
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Contact: <sip:0714****[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1145806395 1145806396 IN IP4 46.182.249.34
s=Asterisk PBX 13.1.0
c=IN IP4 46.182.249.34
t=0 0
m=audio 25052 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK049d19e8;rport=5060
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected]:5062>
User-Agent: DX800A/41.173.00.000.000
Content-Length: 0


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

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK049d19e8;rport=5060
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected]:5062>
User-Agent: DX800A/41.173.00.000.000
User-Agent: DX800A/41.173.00.000.000
Content-Length: 0


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

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK049d19e8;rport=5060
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected]:5062>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 200

v=0
o=6000 8044 42 IN IP4 192.168.178.39
s=Mapping
c=IN IP4 192.168.178.39
t=0 0
m=audio 8044 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<------------->
--- (11 headers 10 lines) ---
Transmitting (NAT) to 192.168.178.39:5062:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK6198d088;rport
Max-Forwards: 70
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Contact: <sip:0714****[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Audio is at 13218
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.178.39:5062:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK1deae232;rport
Max-Forwards: 70
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Contact: <sip:0714****[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1145806395 1145806397 IN IP4 46.182.249.34
s=Asterisk PBX 13.1.0
c=IN IP4 46.182.249.34
t=0 0
m=audio 25052 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=maxptime:150
a=sendrecv

---
Really destroying SIP dialog '[email protected]:5060' Method: BYE

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK1deae232;rport=5060
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Call-ID: [email protected]:5060
CSeq: 104 INVITE
Contact: <sip:[email protected]:5062>
User-Agent: DX800A/41.173.00.000.000
Content-Length: 0


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

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK1deae232;rport=5060
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Call-ID: [email protected]:5060
CSeq: 104 INVITE
Contact: <sip:[email protected]:5062>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 200

v=0
o=6000 8044 43 IN IP4 192.168.178.39
s=Mapping
c=IN IP4 192.168.178.39
t=0 0
m=audio 8044 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<------------->
--- (11 headers 10 lines) ---
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
Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.39:8044
Transmitting (NAT) to 192.168.178.39:5062:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK06d1ad23;rport
Max-Forwards: 70
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Contact: <sip:0714****[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Audio is at 10686
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5083:
INVITE sip:0714****[email protected]:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK7c999e39;rport
Max-Forwards: 70
From: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
To: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 276

v=0
o=root 961197255 961197257 IN IP4 192.168.178.39
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.39
t=0 0
m=audio 8044 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=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK7c999e39;rport=61527
From: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
To: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 103 INVITE
Supported: timer,x-diversion
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0714****[email protected]:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
Require: timer
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk
X-Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY
Session-Expires: 1800;refresher=uac

v=0
o=- 387391714198498277 2 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 25052 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (16 headers 11 lines) ---
Transmitting (NAT) to 46.182.249.38:5083:
ACK sip:0714****[email protected]:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK68a8e066;rport
Max-Forwards: 70
From: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
To: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---

<--- SIP read from UDP:192.168.178.39:5062 --->
INVITE sip:0714****[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bK81300aa2e771118efd5bf2ce313018d2;rport
From: <sip:[email protected]:5062>;tag=3441077112
To: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected]:5062>
Max-Forwards: 70
User-Agent: DX800A/41.173.00.000.000
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 144

v=0
o=6000 8044 44 IN IP4 192.168.178.39
s=Mapping
c=IN IP4 192.168.178.39
t=0 0
m=audio 8044 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sendonly

<------------->
--- (13 headers 8 lines) ---
Sending to 192.168.178.39:5062 (NAT)
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.178.39:8044

<--- Transmitting (NAT) to 192.168.178.39:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bK81300aa2e771118efd5bf2ce313018d2;received=192.168.178.39;rport=5062
From: <sip:[email protected]:5062>;tag=3441077112
To: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0714****[email protected]:5060>
Content-Length: 0


<------------>
Audio is at 13218
Adding codec alaw to SDP
Adding codec ulaw to SDP

<--- Reliably Transmitting (NAT) to 192.168.178.39:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bK81300aa2e771118efd5bf2ce313018d2;received=192.168.178.39;rport=5062
From: <sip:[email protected]:5062>;tag=3441077112
To: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0714****[email protected]:5060>
Content-Type: application/sdp
Content-Length: 221

v=0
o=root 1145806395 1145806398 IN IP4 46.182.249.34
s=Asterisk PBX 13.1.0
c=IN IP4 46.182.249.34
t=0 0
m=audio 25052 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=recvonly

<------------>
Audio is at 10686
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5083:
INVITE sip:0714****[email protected]:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK1594b86d;rport
Max-Forwards: 70
From: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
To: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 961197255 961197258 IN IP4 192.168.178.40
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.40
t=0 0
m=audio 10686 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=maxptime:150
a=sendrecv

---
    -- Started music on hold, class 'default', on channel 'SIP/eqada-00000056'

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK1594b86d;rport=61527
From: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
To: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 104 INVITE
Supported: timer,x-diversion
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0714****[email protected]:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
Require: timer
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk
X-Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY
Session-Expires: 1800;refresher=uac

v=0
o=- 387391714198498277 2 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 25052 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (16 headers 11 lines) ---
Transmitting (NAT) to 46.182.249.38:5083:
ACK sip:0714****[email protected]:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK5a270384;rport
Max-Forwards: 70
From: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
To: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---

<--- SIP read from UDP:192.168.178.39:5062 --->
ACK sip:0714****[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bKb19861a4754726dfcb936966b6730a1b;rport
From: <sip:[email protected]:5062>;tag=3441077112
To: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
Call-ID: [email protected]:5060
CSeq: 103 ACK
Contact: <sip:[email protected]:5062>
Max-Forwards: 70
User-Agent: DX800A/41.173.00.000.000
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3941851115@192_168_178_39' Method: REGISTER

<--- SIP read from UDP:192.168.178.39:5062 --->
INVITE sip:01632****@192.168.178.40;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bKcaf4f4301c4db8806ba40f2f3f045789;rport
From: "rezeption" <sip:[email protected]>;tag=370887398
To: <sip:0163****@192.168.178.40;user=phone>
Call-ID: 3282237177@192_168_178_39
CSeq: 2 INVITE
Contact: <sip:[email protected]:5062>
Max-Forwards: 70
User-Agent: DX800A/41.173.00.000.000
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 373

v=0
o=6000 8046 20 IN IP4 192.168.178.39
s=Mapping
c=IN IP4 192.168.178.39
t=0 0
m=audio 8046 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (14 headers 16 lines) ---
Sending to 192.168.178.39:5062 (NAT)
Sending to 192.168.178.39:5062 (NAT)
Using INVITE request as basis request - 3282237177@192_168_178_39
Found peer '6000' for '6000' from 192.168.178.39:5062
  == Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 96
Found audio description format AAL2-G726-32 for ID 97
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|g726|alaw|g722|g729|g726aal2)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.39:8046
Looking for 0163**** in DLPN_AlleRufe (domain 192.168.178.40)
sip_route_dump: route/path hop: <sip:[email protected]:5062>

<--- Transmitting (NAT) to 192.168.178.39:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bKcaf4f4301c4db8806ba40f2f3f045789;received=192.168.178.39;rport=5062
From: "rezeption" <sip:[email protected]>;tag=370887398
To: <sip:0163****@192.168.178.40;user=phone>
Call-ID: 3282237177@192_168_178_39
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0163****@192.168.178.42:5060>
Content-Length: 0


<------------>
    -- Executing [0163****@DLPN_AlleRufe:1] Macro("SIP/6000-00000058", "trunkdial-failover-0.3,SIP/eqada/0163****,,eqada,") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/6000-00000058", "0?1-fmsetcid,1") in new stack
    -- Executing [[email protected]:2] GotoIf("SIP/6000-00000058", "0?1-setgbobname,1") in new stack
    -- Executing [[email protected]:3] Set("SIP/6000-00000058", "CALLERID(num)=0714****") in new stack
    -- Executing [[email protected]:4] GotoIf("SIP/6000-00000058", "1?1-dial,1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
    -- Executing [[email protected]:1] Dial("SIP/6000-00000058", "SIP/eqada/0163****") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 17504
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5060:
INVITE sip:0163****@15370.pbx-trunk.net SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK016f221d;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 13 Sep 2015 10:01:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 2036384884 2036384884 IN IP4 192.168.178.40
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.40
t=0 0
m=audio 17504 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=maxptime:150
a=sendrecv

---
    -- Called SIP/eqada/0163****

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK016f221d;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>
Call-ID: [email protected]:5060
CSeq: 102 INVITE


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

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK016f221d;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Supported: timer
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Content-Length: 0
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk
Proxy-Authenticate: Digest realm="15370.pbx-trunk.net",nonce="5afedd344ec9e28f2812e2ea45069165",opaque="e735fe172bdb4720b78e7326a8e229f6",algorithm=MD5,qop="auth"


<------------->
--- (12 headers 0 lines) ---
Transmitting (NAT) to 46.182.249.38:5083:
ACK sip:0163****@15370.pbx-trunk.net SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK016f221d;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Audio is at 17504
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5083:
INVITE sip:0163****@15370.pbx-trunk.net SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK1aebc1a1;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="6637d67c256e587f", realm="15370.pbx-trunk.net", algorithm=MD5, uri="sip:0163****@15370.pbx-trunk.net", nonce="5afedd344ec9e28f2812e2ea45069165", response="b322b0d0818182a7bcce81211ead7732", opaque="e735fe172bdb4720b78e7326a8e229f6", qop=auth, cnonce="5e073ace", nc=00000001
Date: Sun, 13 Sep 2015 10:01:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 2036384884 2036384885 IN IP4 192.168.178.40
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.40
t=0 0
m=audio 17504 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=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK1aebc1a1;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>
Call-ID: [email protected]:5060
CSeq: 103 INVITE


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

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK1aebc1a1;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Supported: timer
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0163****@46.182.249.38:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk

v=0
o=- 242115632854794804 1 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 29424 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (13 headers 11 lines) ---
sip_route_dump: route/path hop: <sip:0163****@46.182.249.38:5083;transport=udp>
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
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 46.182.249.34:29424
    -- SIP/eqada-00000059 is making progress passing it to SIP/6000-00000058
Audio is at 10420
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 192.168.178.39:5062 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bKcaf4f4301c4db8806ba40f2f3f045789;received=192.168.178.39;rport=5062
From: "rezeption" <sip:[email protected]>;tag=370887398
To: <sip:0163****@192.168.178.40;user=phone>;tag=as0f536bdc
Call-ID: 3282237177@192_168_178_39
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0163****@192.168.178.42:5060>
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 1207714617 1207714617 IN IP4 192.168.178.42
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.42
t=0 0
m=audio 10420 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=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK1aebc1a1;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Supported: timer
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0163****@46.182.249.38:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk

v=0
o=- 242115632854794804 1 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 29424 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (13 headers 11 lines) ---
sip_route_dump: route/path hop: <sip:0163****@46.182.249.38:5083;transport=udp>
    -- SIP/eqada-00000059 is ringing
    -- SIP/eqada-00000059 is making progress passing it to SIP/6000-00000058
       > 0x2b47e138 -- Probation passed - setting RTP source address to 192.168.178.39:8046
       > 0x2b7db9b8 -- Probation passed - setting RTP source address to 46.182.249.34:29424

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK1aebc1a1;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Supported: timer,x-diversion
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0163****@46.182.249.38:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk
X-Call-ID: [email protected]:5060-UASession-*QvODzM5Mn

v=0
o=- 242115632854794804 1 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 29424 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (14 headers 11 lines) ---
sip_route_dump: route/path hop: <sip:0163****@46.182.249.38:5083;transport=udp>
Transmitting (NAT) to 46.182.249.38:5083:
ACK sip:0163****@46.182.249.38:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK7aa58905;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
    -- SIP/eqada-00000059 answered SIP/6000-00000058
Audio is at 10420
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.178.39:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bKcaf4f4301c4db8806ba40f2f3f045789;received=192.168.178.39;rport=5062
From: "rezeption" <sip:[email protected]>;tag=370887398
To: <sip:0163****@192.168.178.40;user=phone>;tag=as0f536bdc
Call-ID: 3282237177@192_168_178_39
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0163****@192.168.178.42:5060>
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 1207714617 1207714617 IN IP4 192.168.178.42
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.42
t=0 0
m=audio 10420 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=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/6000-00000058 joined 'simple_bridge' basic-bridge <8f627339-0de0-485f-aee8-ba055813077c>
    -- Channel SIP/eqada-00000059 joined 'simple_bridge' basic-bridge <8f627339-0de0-485f-aee8-ba055813077c>
       > Bridge 8f627339-0de0-485f-aee8-ba055813077c: switching from simple_bridge technology to native_rtp
Audio is at 17504
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5083:
INVITE sip:0163****@46.182.249.38:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK163cdbc1;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 2036384884 2036384886 IN IP4 192.168.178.39
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.39
t=0 0
m=audio 8046 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=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:192.168.178.39:5062 --->
ACK sip:0163****@192.168.178.42:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bK5892a342d5ce46a7c21944854624eba8;rport
From: "rezeption" <sip:[email protected]>;tag=370887398
To: <sip:0163****@192.168.178.40;user=phone>;tag=as0f536bdc
Call-ID: 3282237177@192_168_178_39
CSeq: 2 ACK
Contact: <sip:[email protected]:5062>
Max-Forwards: 70
User-Agent: DX800A/41.173.00.000.000
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Audio is at 10420
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.178.39:5062:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK67e62821;rport
Max-Forwards: 70
From: <sip:0163****@192.168.178.40;user=phone>;tag=as0f536bdc
To: "rezeption" <sip:[email protected]>;tag=370887398
Contact: <sip:0163****@192.168.178.42:5060>
Call-ID: 3282237177@192_168_178_39
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1207714617 1207714618 IN IP4 46.182.249.34
s=Asterisk PBX 13.1.0
c=IN IP4 46.182.249.34
t=0 0
m=audio 29424 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=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK163cdbc1;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Call-ID: [email protected]:5060
CSeq: 104 INVITE
Supported: timer,x-diversion
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0163****@46.182.249.38:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk
X-Call-ID: [email protected]:5060-UASession-*QvODzM5Mn

v=0
o=- 242115632854794804 1 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 29424 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (14 headers 11 lines) ---
Transmitting (NAT) to 46.182.249.38:5083:
ACK sip:0163****@46.182.249.38:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK64ec4eda;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---

<--- SIP read from UDP:192.168.178.20:5060 --->
REGISTER sip:192.168.178.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.20:5060;branch=z9hG4bK00eba80b6c58e511b2a68631643d752c;rport
From: <sip:[email protected]>;tag=1161568889
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 52 REGISTER
Contact: <sip:[email protected]:5060>;+sip.instance="<urn:uuid:00DAB3F9-3737-E411-9061-2C64F20B5AF3>"
Authorization: Digest username="6070", realm="15370.pbx-trunk.net", nonce="7c436386", uri="sip:192.168.178.40", response="a68a695e860ed4f5c4e6e5b3d9c3f2dc", algorithm=MD5
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Expires: 600
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.178.20:5060 (NAT)
Sending to 192.168.178.20:5060 (NAT)

<--- Transmitting (NAT) to 192.168.178.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.20:5060;branch=z9hG4bK00eba80b6c58e511b2a68631643d752c;received=192.168.178.20;rport=5060
From: <sip:[email protected]>;tag=1161568889
To: <sip:[email protected]>;tag=as43676430
Call-ID: [email protected]
CSeq: 52 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="15370.pbx-trunk.net", nonce="096d2ca4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.178.20:5060 --->
REGISTER sip:192.168.178.40 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.20:5060;branch=z9hG4bK00eba80b6c58e511b2a78631643d752c;rport
From: <sip:[email protected]>;tag=1161568889
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 53 REGISTER
Contact: <sip:[email protected]:5060>;+sip.instance="<urn:uuid:00DAB3F9-3737-E411-9061-2C64F20B5AF3>"
Authorization: Digest username="6070", realm="15370.pbx-trunk.net", nonce="096d2ca4", uri="sip:192.168.178.40", response="d103fcb28cb72ad75332052da4923dd3", algorithm=MD5
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Expires: 600
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.178.20:5060 (NAT)

<--- Transmitting (NAT) to 192.168.178.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.20:5060;branch=z9hG4bK00eba80b6c58e511b2a78631643d752c;received=192.168.178.20;rport=5060
From: <sip:[email protected]>;tag=1161568889
To: <sip:[email protected]>;tag=as43676430
Call-ID: [email protected]
CSeq: 53 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:[email protected]:5060>;expires=600
Date: Sun, 13 Sep 2015 10:01:17 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.178.20:5060:
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK33c2a090;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as1ebd6460
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 94

Messages-Waiting: no
Message-Account: sip:[email protected]
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.178.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK33c2a090;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as1ebd6460
To: <sip:[email protected]:5060>;tag=00eba80b6c58e511b2a88631643d752c
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
Contact: <sip:[email protected]:5060>
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Server: SIPPER for PhonerLite
Content-Length: 0


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

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK67e62821;rport=5060
From: <sip:0163****@192.168.178.40;user=phone>;tag=as0f536bdc
To: "rezeption" <sip:[email protected]>;tag=370887398
Call-ID: 3282237177@192_168_178_39
CSeq: 102 INVITE
Contact: <sip:[email protected]:5062>
User-Agent: DX800A/41.173.00.000.000
Content-Length: 0


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

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK67e62821;rport=5060
From: <sip:0163****@192.168.178.40;user=phone>;tag=as0f536bdc
To: "rezeption" <sip:[email protected]>;tag=370887398
Call-ID: 3282237177@192_168_178_39
CSeq: 102 INVITE
Contact: <sip:[email protected]:5062>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 200

v=0
o=6000 8046 21 IN IP4 192.168.178.39
s=Mapping
c=IN IP4 192.168.178.39
t=0 0
m=audio 8046 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<------------->
--- (11 headers 10 lines) ---
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
Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.39:8046
Transmitting (NAT) to 192.168.178.39:5062:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK3fd672a9;rport
Max-Forwards: 70
From: <sip:0163****@192.168.178.40;user=phone>;tag=as0f536bdc
To: "rezeption" <sip:[email protected]>;tag=370887398
Contact: <sip:0163****@192.168.178.42:5060>
Call-ID: 3282237177@192_168_178_39
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Audio is at 17504
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5083:
INVITE sip:0163****@46.182.249.38:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK5a7be40b;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 2036384884 2036384887 IN IP4 192.168.178.39
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.39
t=0 0
m=audio 8046 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=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK5a7be40b;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Call-ID: [email protected]:5060
CSeq: 105 INVITE
Supported: timer,x-diversion
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0163****@46.182.249.38:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk
X-Call-ID: [email protected]:5060-UASession-*QvODzM5Mn

v=0
o=- 242115632854794804 1 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 29424 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (14 headers 11 lines) ---
Transmitting (NAT) to 46.182.249.38:5083:
ACK sip:0163****@46.182.249.38:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK019a68f2;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 105 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---

<--- SIP read from UDP:192.168.178.39:5062 --->
REFER sip:0714****[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bKcd200d3fecddb752dfe4396afc8d297f;rport
From: <sip:[email protected]:5062>;tag=3441077112
To: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
Call-ID: [email protected]:5060
CSeq: 104 REFER
Contact: <sip:[email protected]:5062>
Max-Forwards: 70
User-Agent: DX800A/41.173.00.000.000
Supported: replaces
Refer-To: <sip:0163****@192.168.178.42:5060?Replaces=3282237177%40192_168_178_39%3Bfrom-tag%3D370887398%3Bto-tag%3Das0f536bdc>
Referred-By: <sip:[email protected]>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Call [email protected]:5060 got a SIP call transfer from caller: (REFER)!
SIP transfer to extension 0163****@DLPN_AlleRufe by [email protected]

<--- Transmitting (NAT) to 192.168.178.39:5062 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bKcd200d3fecddb752dfe4396afc8d297f;received=192.168.178.39;rport=5062
From: <sip:[email protected]:5062>;tag=3441077112
To: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
Call-ID: [email protected]:5060
CSeq: 104 REFER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0714****[email protected]:5060>
Content-Length: 0


<------------>
    -- Channel SIP/eqada-00000056 left 'native_rtp' basic-bridge <d3518699-4807-4bd1-8f92-3d5c4094b801>
    -- Channel SIP/eqada-00000056 swapped with SIP/6000-00000058 into 'native_rtp' basic-bridge <8f627339-0de0-485f-aee8-ba055813077c>
    -- Channel SIP/6000-00000058 left 'native_rtp' basic-bridge <8f627339-0de0-485f-aee8-ba055813077c>
Audio is at 17504
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5083:
INVITE sip:0163****@46.182.249.38:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK3b9e8e42;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 106 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 2036384884 2036384888 IN IP4 192.168.178.40
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.40
t=0 0
m=audio 17504 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=maxptime:150
a=sendrecv

---
Audio is at 10686
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5083:
INVITE sip:0714****[email protected]:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK00430b69;rport
Max-Forwards: 70
From: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
To: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 275

v=0
o=root 961197255 961197259 IN IP4 46.182.249.34
s=Asterisk PBX 13.1.0
c=IN IP4 46.182.249.34
t=0 0
m=audio 29424 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=maxptime:150
a=sendrecv

---
       > Bridge d3518699-4807-4bd1-8f92-3d5c4094b801: switching from native_rtp technology to simple_bridge
    -- Channel SIP/6000-00000057 left 'simple_bridge' basic-bridge <d3518699-4807-4bd1-8f92-3d5c4094b801>
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: REFER)
  == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/6000-00000058' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_AlleRufe, 0163****, 1) exited non-zero on 'SIP/6000-00000058'
Scheduling destruction of SIP dialog '3282237177@192_168_178_39' in 32000 ms (Method: ACK)
Reliably Transmitting (NAT) to 192.168.178.39:5062:
BYE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK5b82d91b;rport
Max-Forwards: 70
From: <sip:0163****@192.168.178.40;user=phone>;tag=as0f536bdc
To: "rezeption" <sip:[email protected]>;tag=370887398
Call-ID: 3282237177@192_168_178_39
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Reliably Transmitting (NAT) to 192.168.178.39:5062:
NOTIFY sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK55139a21;rport
Max-Forwards: 70
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Contact: <sip:0714****[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX
Event: refer;id=104
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 16

SIP/2.0 200 OK

---
    -- Stopped music on hold on SIP/eqada-00000056

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK00430b69;rport=61527
From: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
To: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 105 INVITE
Supported: timer,x-diversion
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0714****[email protected]:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
Require: timer
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk
X-Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY
Session-Expires: 1800;refresher=uac

v=0
o=- 387391714198498277 2 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 25052 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (16 headers 11 lines) ---
Transmitting (NAT) to 46.182.249.38:5083:
ACK sip:0714****[email protected]:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK6eac01e7;rport
Max-Forwards: 70
From: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as02503cba
To: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Fu00YNCK30HNQ8BK
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: 8415e00015f5-55f5494d-3e6228a9-1259c100-53ab31@127.0.0.1-UASession-D076JRKsNY-UASession-UNGjdLKUgx
CSeq: 105 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK3b9e8e42;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Call-ID: [email protected]:5060
CSeq: 106 INVITE
Supported: timer,x-diversion
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0163****@46.182.249.38:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk
X-Call-ID: [email protected]:5060-UASession-*QvODzM5Mn

v=0
o=- 242115632854794804 1 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 29424 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (14 headers 11 lines) ---
Transmitting (NAT) to 46.182.249.38:5083:
ACK sip:0163****@46.182.249.38:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK04c64010;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 106 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Audio is at 17504
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5083:
INVITE sip:0163****@46.182.249.38:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK29bc4346;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 107 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 2036384884 2036384889 IN IP4 46.182.249.34
s=Asterisk PBX 13.1.0
c=IN IP4 46.182.249.34
t=0 0
m=audio 25052 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=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK29bc4346;rport=61527
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Call-ID: [email protected]:5060
CSeq: 107 INVITE
Supported: timer,x-diversion
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0163****@46.182.249.38:5083;transport=udp>
Content-Length: 230
Content-Type: application/sdp
User-Agent: TELES.C5/5.0.7.8
Allow-Events: talk
X-Call-ID: [email protected]:5060-UASession-*QvODzM5Mn

v=0
o=- 242115632854794804 1 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 29424 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -

<------------->
--- (14 headers 11 lines) ---
Transmitting (NAT) to 46.182.249.38:5083:
ACK sip:0163****@46.182.249.38:5083;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK10058f5f;rport
Max-Forwards: 70
From: "Rezeption" <sip:0714****@192.168.178.40>;tag=as1fe70930
To: <sip:0163****@15370.pbx-trunk.net>;tag=aAI6u_sG-c
Contact: <sip:0714****@192.168.178.40:5060>
Call-ID: [email protected]:5060
CSeq: 107 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK5b82d91b;rport=5060
From: <sip:0163****@192.168.178.40;user=phone>;tag=as0f536bdc
To: "rezeption" <sip:[email protected]>;tag=370887398
Call-ID: 3282237177@192_168_178_39
CSeq: 103 BYE
Contact: <sip:[email protected]:5062>
Supported: replaces
User-Agent: DX800A/41.173.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '3282237177@192_168_178_39' Method: ACK

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK55139a21;rport=5060
From: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
To: <sip:[email protected]:5062>;tag=3441077112
Call-ID: [email protected]:5060
CSeq: 105 NOTIFY
Contact: <sip:[email protected]:5062>
Supported: replaces
User-Agent: DX800A/41.173.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


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

<--- SIP read from UDP:192.168.178.39:5062 --->
BYE sip:0714****[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bK51e264237a1fc125e868146b95153fa7;rport
From: <sip:[email protected]:5062>;tag=3441077112
To: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
Call-ID: [email protected]:5060
CSeq: 105 BYE
Contact: <sip:[email protected]:5062>
Max-Forwards: 70
User-Agent: DX800A/41.173.00.000.000
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.178.39:5062 (NAT)
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.178.39:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.39:5062;branch=z9hG4bK51e264237a1fc125e868146b95153fa7;received=192.168.178.39;rport=5062
From: <sip:[email protected]:5062>;tag=3441077112
To: "0714****5608" <sip:0714****[email protected]>;tag=as0ee31a52
Call-ID: [email protected]:5060
CSeq: 105 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[Sep 13 12:01:24] NOTICE[18052]: chan_sip.c:15178 sip_reregister:    -- Re-registration for  [EMAIL="[email protected]"][email protected][/EMAIL]
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 46.182.249.38:5060:
REGISTER sip:15370.pbx-trunk.net SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK3cf7d2cc;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as25e37051
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1093 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX
Authorization: Digest username="6637d67c256e587f", realm="15370.pbx-trunk.net", algorithm=MD5, uri="sip:15370.pbx-trunk.net", nonce="594d13d5415dd96e06b43a864b42a82a", response="eb55f29337c61ec52a259eb181c153b1", opaque="d55a0e753d290b66d668c8a32c7c098d", qop=auth, cnonce="007cbcba", nc=00000014
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0


---

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK3cf7d2cc;rport=61527
From: <sip:[email protected]>;tag=as25e37051
To: <sip:[email protected]>;tag=aprqkf8psp3-sn1g8o00001a4
Call-ID: [email protected]
CSeq: 1093 REGISTER
Contact: <sip:[email protected]:5060>;expires=60


<------------->
--- (7 headers 0 lines) ---
[Sep 13 12:01:24] NOTICE[18052]: chan_sip.c:23747 handle_response_register: Outbound Registration: Expiry for 15370.pbx-trunk.net is 60 sec (Scheduling reregistration in 45 s)
Really destroying SIP dialog '[email protected]' Method: REGISTER
Cronos_Asterisk*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
Cronos_Asterisk>

Cronos_Asterisk>
Cronos_Asterisk>
Cronos_Asterisk> ./asterisk -vvvvvr
Asterisk 13.1.0, Copyright (C) 1999 - 2014, 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.
=========================================================================
Connected to Asterisk 13.1.0 currently running on Cronos_Asterisk (pid = 17995)

<--- SIP read from UDP:46.182.249.38:5083 --->
INVITE sip:0714****@192.168.178.40:5060 SIP/2.0
Via: SIP/2.0/UDP 46.182.249.38:5083;branch=z9hG4bKgkm1ir100go1plge71q0.1
From: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Xu002YRAH1HWNY05
To: <sip:0714****@192.168.2.80:5060;transport=udp>
Call-ID: [EMAIL="8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU"]8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU[/EMAIL]8vvM-UASession-hq4GVDXybJ
CSeq: 1 INVITE
Max-Forwards: 68
Supported: timer
Unsupported: refer
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:0714****[email protected]:5083;transport=udp>
Content-Length: 242
Content-Type: application/sdp
User-Agent: TELES.MGC
Allow-Events: talk
Accept: application/sdp
P-Charging-Vector: icid-value=PoS-1442139282
Privacy: none
X-IP-Info: 192.168.2.106

v=0
o=- 242936182488777523 1 IN IP4 46.182.249.34
s=-
c=IN IP4 46.182.249.34
t=0 0
m=audio 29780 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -

<------------->
--- (19 headers 11 lines) ---
Sending to 46.182.249.38:5083 (NAT)
Sending to 46.182.249.38:5083 (NAT)
Using INVITE request as basis request - [EMAIL="8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU"]8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU[/EMAIL]8vvM-UASession-hq4GVDXybJ
Found peer 'eqada' for '0714****5608' from 46.182.249.38:5083
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
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 telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 46.182.249.34:29780
Looking for 0714**** in DID_eqada (domain 192.168.178.40)
sip_route_dump: route/path hop: <sip:0714****[email protected]:5083;transport=udp>

<--- Transmitting (NAT) to 46.182.249.38:5083 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 46.182.249.38:5083;branch=z9hG4bKgkm1ir100go1plge71q0.1;received=46.182.249.38;rport=5083
From: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Xu002YRAH1HWNY05
To: <sip:0714****@192.168.2.80:5060;transport=udp>
Call-ID: [EMAIL="8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU"]8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU[/EMAIL]8vvM-UASession-hq4GVDXybJ
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:0714****@192.168.178.40:5060>
Content-Length: 0


<------------>
    -- Executing [0714****@DID_eqada:1] Goto("SIP/eqada-0000005f", "default,6000,1") in new stack
    -- Goto (default,6000,1)
    -- Executing [6000@default:1] Dial("SIP/eqada-0000005f", "SIP/6000") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 13670
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.178.39:5062:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK633ddeb1;rport
Max-Forwards: 70
From: "0714****5608" <sip:0714****[email protected]>;tag=as759fff98
To: <sip:[email protected]:5062>
Contact: <sip:0714****[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 13 Sep 2015 10:14:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 1948849112 1948849112 IN IP4 192.168.178.42
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.42
t=0 0
m=audio 13670 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/6000

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK633ddeb1;rport=5060
From: "0714****5608" <sip:0714****[email protected]>;tag=as759fff98
To: <sip:[email protected]:5062>;tag=696978327
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected]:5062>
User-Agent: DX800A/41.173.00.000.000
Content-Length: 0


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

<--- SIP read from UDP:192.168.178.39:5062 --->
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK633ddeb1;rport=5060
From: "0714****5608" <sip:0714****[email protected]>;tag=as759fff98
To: <sip:[email protected]:5062>;tag=696978327
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:0163****@192.168.178.40;user=phone>
User-Agent: DX800A/41.173.00.000.000
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- Got SIP response 302 "Moved Temporarily" back from 192.168.178.39:5062
Transmitting (NAT) to 192.168.178.39:5062:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;branch=z9hG4bK633ddeb1;rport
Max-Forwards: 70
From: "0714****5608" <sip:0714****[email protected]>;tag=as759fff98
To: <sip:[email protected]:5062>;tag=696978327
Contact: <sip:0714****[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
    -- Now forwarding SIP/eqada-0000005f to 'Local/0163****@DLPN_AlleRufe' (thanks to SIP/6000-00000060)
[Sep 13 12:14:42] NOTICE[18597][C-00000068]: app_dial.c:891 do_forward: Not accepting call completion offers from call-forward recipient Local/0163****@DLPN_AlleRufe-00000047;1

<--- Transmitting (NAT) to 46.182.249.38:5083 --->
SIP/2.0 181 Call is being forwarded
Via: SIP/2.0/UDP 46.182.249.38:5083;branch=z9hG4bKgkm1ir100go1plge71q0.1;received=46.182.249.38;rport=5083
From: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Xu002YRAH1HWNY05
To: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as0087825c
Call-ID: [EMAIL="8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU"]8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU[/EMAIL]8vvM-UASession-hq4GVDXybJ
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:0714****@192.168.178.40:5060>
Diversion: <sip:[email protected]>;reason=unconditional
Content-Length: 0


<------------>
    -- Executing [0163****@DLPN_AlleRufe:1] Macro("Local/0163****@DLPN_AlleRufe-00000047;2", "trunkdial-failover-0.3,SIP/eqada/0163****,,eqada,") in new stack
    -- Executing [[email protected]:1] GotoIf("Local/0163****@DLPN_AlleRufe-00000047;2", "0?1-fmsetcid,1") in new stack
    -- Executing [[email protected]:2] GotoIf("Local/0163****@DLPN_AlleRufe-00000047;2", "0?1-setgbobname,1") in new stack
    -- Executing [[email protected]:3] Set("Local/0163****@DLPN_AlleRufe-00000047;2", "CALLERID(num)=") in new stack
    -- Executing [[email protected]:4] GotoIf("Local/0163****@DLPN_AlleRufe-00000047;2", "0?1-dial,1") in new stack
    -- Executing [[email protected]:5] Set("Local/0163****@DLPN_AlleRufe-00000047;2", "CALLERID(all)=") in new stack
    -- Executing [[email protected]:6] SIPAddHeader("Local/0163****@DLPN_AlleRufe-00000047;2", "P-Preferred-Identity:<sip:@15370.pbx-trunk.net>") in new stack
    -- Executing [[email protected]:7] Goto("Local/0163****@DLPN_AlleRufe-00000047;2", "1-dial,1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
    -- Executing [[email protected]:1] Dial("Local/0163****@DLPN_AlleRufe-00000047;2", "SIP/eqada/0163****") in new stack
  == Using SIP RTP CoS mark 5
Really destroying SIP dialog '[email protected]:5060' Method: INVITE
Audio is at 13624
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 46.182.249.38:5060:
INVITE sip:0163****@15370.pbx-trunk.net SIP/2.0
Via: SIP/2.0/UDP 192.168.178.40:5060;branch=z9hG4bK3e2685fa;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as2965f102
To: <sip:0163****@15370.pbx-trunk.net>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 13 Sep 2015 10:14:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Preferred-Identity: <sip:@15370.pbx-trunk.net>
Diversion: <sip:[email protected]>;reason=unconditional
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 240461276 240461276 IN IP4 192.168.178.40
s=Asterisk PBX 13.1.0
c=IN IP4 192.168.178.40
t=0 0
m=audio 13624 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/eqada/0163****

<--- SIP read from UDP:46.182.249.38:5083 --->
SIP/2.0 400 Invalid P-Preferred-Identity
Via: SIP/2.0/UDP 192.168.178.40:5060;received=79.219.206.152;branch=z9hG4bK3e2685fa;rport=61527
From: "asterisk" <sip:[email protected]>;tag=as2965f102
To: <sip:0163****@15370.pbx-trunk.net>;tag=aprqngfrt-m8n5dm00000c6
Call-ID: [email protected]:5060
CSeq: 102 INVITE


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

<--- SIP read from UDP:46.182.249.38:5083 --->
CANCEL sip:0714****@192.168.178.40:5060 SIP/2.0
Via: SIP/2.0/UDP 46.182.249.38:5083;branch=z9hG4bKgkm1ir100go1plge71q0.1
CSeq: 1 CANCEL
From: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Xu002YRAH1HWNY05
To: <sip:0714****@192.168.2.80:5060;transport=udp>
Call-ID: [EMAIL="8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU"]8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU[/EMAIL]8vvM-UASession-hq4GVDXybJ
Max-Forwards: 68
Content-Length: 0
Reason: Q.850;cause=16;text="Normal call clearing"


<------------->
--- (9 headers 0 lines) ---
Sending to 46.182.249.38:5083 (NAT)

<--- Reliably Transmitting (NAT) to 46.182.249.38:5083 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 46.182.249.38:5083;branch=z9hG4bKgkm1ir100go1plge71q0.1;received=46.182.249.38;rport=5083
From: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Xu002YRAH1HWNY05
To: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as0087825c
Call-ID: [EMAIL="8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU"]8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU[/EMAIL]8vvM-UASession-hq4GVDXybJ
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (NAT) to 46.182.249.38:5083 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 46.182.249.38:5083;branch=z9hG4bKgkm1ir100go1plge71q0.1;received=46.182.249.38;rport=5083
From: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Xu002YRAH1HWNY05
To: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as0087825c
Call-ID: [EMAIL="8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU"]8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU[/EMAIL]8vvM-UASession-hq4GVDXybJ
CSeq: 1 CANCEL
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
  == Spawn extension (default, 6000, 1) exited non-zero on 'SIP/eqada-0000005f'
  == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'Local/0163****@DLPN_AlleRufe-00000047;2' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_AlleRufe, 0163****, 1) exited non-zero on 'Local/0163****@DLPN_AlleRufe-00000047;2'

<--- SIP read from UDP:46.182.249.38:5083 --->
ACK sip:0714****@192.168.178.40:5060 SIP/2.0
Via: SIP/2.0/UDP 46.182.249.38:5083;branch=z9hG4bKgkm1ir100go1plge71q0.1
CSeq: 1 ACK
From: "0714****5608" <sip:0714****[email protected]>;tag=0UUHS0000030000E1D0100Xu002YRAH1HWNY05
To: <sip:0714****@192.168.2.80:5060;transport=udp>;tag=as0087825c
Call-ID: [EMAIL="8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU"]8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU[/EMAIL]8vvM-UASession-hq4GVDXybJ
Max-Forwards: 68
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '8415e00015f5-55f54c92-2cebddc6-2b9516d8-53abbd@127.0.0.1-UASession-1d.lxU8vvM-UASession-hq4GVDXybJ' Method: ACK
Cronos_Asterisk*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
Cronos_Asterisk>
 
Zuletzt bearbeitet:
Das Protokoll kannst du gerne in code - Tags klammern.

Welche Firmware läuft auf dem Gigaset ?
 
Sorry wegen den Tags. Ich übe noch....
Das DX800A ist auf V41.00-173.00.00
 
Nun lasst mich nicht im Stich! :(
Hat noch niemand eine AWS an einem Gigaset realisiert?? :lamer:
 
Am Ende von den drei Metern Log findet sich dann doch noch die entscheidende Stelle.

Die Umleitung wird eigentlich sauber ausgeführt, aber der Provider lehnt den Rufaufbau zum Umleitungsziel mit SIP/2.0 400 Invalid P-Preferred-Identity ab.
Stimmt auch, denn P-Preferred-Identity: <sip:mad:15370.pbx-trunk.net> ist ungültig.

Das sieht nach einem GUI powered Asterisk aus, richtig? Dann müsstest Du Dich am besten direkt an den Hersteller wenden, da muss man vermutlich direkt an die Dialplanvorlagen ran, damit der Header entweder richtig befüllt oder erst gar nicht gesendet wird.
 
Das wars! Vielen Dank!

Es hatte immer von intern funktioniert. Da hier die CALLERID(num) gesetzt wird.
Bei externen Anrufen wurde/wird die CALLERID(num) nicht gesetzt (warum muss ich noch prüfen).
Ich habe nun die CALLERID(all) gesetzt und nun funktioniert zumindest die AWS.
Momentan wird auf meinem Handy das angezeigt, was in CALLERID(all) steht und nicht die CALLERID des Anrufers.

Falls ich das Problem mit der CALLERID lösen kann, poste ich noch mal.

Das sieht nach einem GUI powered Asterisk aus, richtig?
Zum Vergrößern anklicken....
Korrekt. Synology.
Asterisk/13.1.0
Asterisk GUI-version : 2.1.0-rc1
 
Die Callerid des Anrufers kannst Du seitens Asterisk nur übertragen, wenn Dein Provider clip-no-screen unterstützt. Besser wäre es, so wie die Fritzbox das auch macht, ein Redirect (SIP 302) zu schicken.
 
Wenn das zu aktivierende Weiterleitungsziel immer Dein Handy sein soll, kannst Du für die Weiterleitung auch z.B. auch cheapvoip nehmen. Darüber kannst Du per sip-uri: [email protected] per SIP-Uri erreicht werden. Setzt Du bei cheapvoip einmalig mit dem Windows-Client von Cheapvoip eine Rufumleitung für alle eingehenden Anrufe auf Deine Handynummer, so kannst Du im Asterisk die Weiterleitung aufs Handy durch Weiterleitung auf diese Sip-URI ersetzen. Da Du bei SIP-Calls die Absendenummer frei setzen kannst, kannst Du so die Anrufernummer als Absendenummer für den SIP-Call verwenden. Die bei Cheapvoip gesetzte Rufumleitung behält diese Nummer bei.
Nur Sonderrufnummern kann man auf diese Weise nicht ans Handy übertragen.

Ganz nebenbei ist das auch der wohl billigste Weg (ca. 0,7 cent/min)

Ich selbst aktiviere und deaktiviere die Rufumleitung gar nicht erst, sondern lasse den Asterisk parallel zu hause, im Büro, auf dem Handy, auf dem Smartphone und der Smartwatch klingeln. Wichtig ist nur, dass die Mobilgeräte keine Nichterreichbarkeitsansage liefern, weil die Ansage als Anrufannahme gewertet würden. Ich löse das durch eine ansonsten ungenutzte Nummer, bei der der Asterisk ring() liefert. AB muss ich dazu zwar am Handy abschalten, da ich aber niemanden die Handynummern gegeben habe (bin ja überall mit der Festnetznummer erreichbar) ist das kein Problem.
 
Hallole,
ersmal vielen Dank für die zwei Tipps.

Unser Provider (equada) unterstütz clip-no-screen. Das funktioniert auch tadellos,
wenn ich in Asterisk eine Rufumleitung programmiere (aktiviere).
Nur die im Gigaset DX800A eingebaute AWS bricht ab, da die CallerId nicht gesetzt ist.

Ich habe das auch mal mit PhonerLite getestet: selbes Verhalten. Es wird keine CallerId gezogen.
Nur wenn ich die CallerId(all) in der SIP.conf setze wird diese angezeigt.
Das mit dem Redirect (SIP 302) muss ich mir mal ganz in Ruhe zur Gemüte führen.....:bluescre:

Ganz nebenbei ist das auch der wohl billigste Weg (ca. 0,7 cent/min)
Zum Vergrößern anklicken....
Wir berappen 0,0882 Eroner bei equada.
 
Das Problem ist, dass Asterisk den Anruf an das Telefon weiter gibt, von dort ein Redirect bekommt, und daraufhin über einen zweiten Channel den abgehenden Anruf für die Umleitung aufbaut. Das ganze läuft über Local so ab, als hätte das Telefon die externe Nummer gewählt, als mit dessen Einstellungen einschließlich Callerid.

Entweder müsste man die Umleitung auswerten und eben das Redirect quasi weiterreichen, das hab ich mal in Gemeinschaft 3 umgebastelt, da ist der Mechanismus nämlich ähnlich.
Oder man kennzeichnet den Anruf mit einer vererbbaren Channel-Variable als weitergeleitet, damit man davon abhängig später die Callerid des Local-Channels überschreiben kann.

Beides geht nur durch Eingriffe in den Dialplan, was bei GUI ja bekanntlich nicht so einfach geht. Da muss man an die Vorlagen dran, aus denen das GUI die extensions.conf erzeugt.
 
Hmmmm....
:bahnhof:
Im Log steht:
"Got SIP response 302 "Moved Temporarily" back from 192.168.178.39:5062"
Das sollte doch der Redirect sein oder??

Geht trotzdem nicht. CallerID ist leer und die Warnung:
"NOTICE[7825][C-00000017]: app_dial.c:891 do_forward: Not accepting call completion offers from call-forward recipient"
kommt ebenfalls noch.

Ich geb's auf.
Dann wird halt immer unsere zentrale Rufnummer auf'm Handy angezeigt.
Somit weiß ich auch gleich, ob es ein umgeleiteter Anruf ist oder ein Direktanruf.

P.s.: An die Vorlage (PBX.js) für die extentions.conf musste ich schon mal ran, da die
P-Preferred-Identity nicht gesetzt wurde. Habe ich (Gottseindank) über einen Hinweis fixen können.
 
Zuletzt bearbeitet:
Da steht "Got SIP response 302", das heißt Asterisk hat ein 302 vom Telefon empfangen. Das heißt aber noch lange nicht, dass er es auch an den Provider weiterleitet.

An die Vorlage (PBX.js) für die extentions.conf musste ich schon mal ran, da die P-Preferred-Identity nicht gesetzt wurde.
Zum Vergrößern anklicken....

Aha, genau das ist nämlich jetzt die Ursache für den ursprünglichen Fehler. Der Header wird bei einem weitergeleitet Anruf eben falsch gesetzt. Da war wohl der Fix nicht ganz bis zum Ende gedacht.

Das könnte man abfangen, in dem man bei ankommenden Anrufen vor dem Dial ein Set(__ankommend=1) macht und das SipAddHeader(P-Preferred-Identity...) im Falle einer Weiterleitung mit ExecIf($["${ankommend}"!="1"]?SipAddHeader(P-Preferred-Identity...)) ausschließt.
 
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