[erledigt]kein Dial nach Record

[boelkstoff]

Hallo,
ich versuche gerade ein internes GEspräch mittels RECORD aufzunehmen:

exten => _000XX,1,Record(asterisk-recording%d:gsm)
exten => _000XX,n,Dial,SIP/${EXTEN}|55|tTrL(3600000)  die Rufdauer
exten => _000XX,n,Goto,r-${DIALSTATUS}|1  
exten => r-BUSY,1,Hangup
exten => r-CONGESTION,1,Hangup
exten => r-CHANUNAVAIL,1,Hangup
exten => r-NOANSWER,1,Hangup

Leider höre ich nur den Piep Ton und danach wird nicht weiterverbunden.

Hier einen Teil der Debugmeldungen (der Anfang fehlt):

Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length:   373

v=0
o=user 151697 151697 IN IP4 85.181.72.90
s=call
c=IN IP4 85.181.72.90
t=1181581106 1181584706
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:7079

--- (17 headers 16 lines) ---
Using INVITE request as basis request - [email protected]
Sending to 85.181.72.90 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 85.181.72.90:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 85.181.72.90:5060;branch=z9hG4bK7C5554E57B9B9D20;received=85.181.72.90
From: <sip:[email protected]>;tag=22F223A07CE0FC39
To: <sip:[email protected]>;tag=as6d63ccaa
Call-ID: [email protected]
CSeq: 3672 INVITE
User-Agent: FRITZ!Box Fon WLAN 7050 Firmware Version 14.04.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35b76022"
Content-Length: 0


---
Scheduling destruction of call '[email protected]' in 15000 ms
Found user '00013'
vs1128*CLI>
<-- SIP read from 85.181.72.90:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 85.181.72.90:5060;branch=z9hG4bK7C5554E57B9B9D20
From: <sip:[email protected]>;tag=22F223A07CE0FC39
To: <sip:[email protected]>;tag=as6d63ccaa
Call-ID: [email protected]
CSeq: 3672 ACK
User-Agent: FRITZ!Box Fon Eumex300IP 15.04.27 (Sep  6 2006)
Content-Length: 0


--- (8 headers 0 lines) ---
vs1128*CLI>
<-- SIP read from 85.181.72.90:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 85.181.72.90:5060;branch=z9hG4bK8BED09BA1E2FE891
From: <sip:[email protected]>;tag=22F223A07CE0FC39
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 3673 INVITE
Contact: <sip:[email protected];uniq=DA342FCCC433313B8A3A90FFD5BA1>
Proxy-Authorization: Digest username="00013", realm="asterisk", nonce="35b76022", uri="sip:[email protected]", response="c8b52942b7f58c6ef6f5c5ed8c75eebf", algorithm=MD5
Max-Forwards: 70
Expires: 120
User-Agent: FRITZ!Box Fon Eumex300IP 15.04.27 (Sep  6 2006)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length:   373

v=0
o=user 151697 151697 IN IP4 85.181.72.90
s=call
c=IN IP4 85.181.72.90
t=1181581106 1181584706
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:7079

--- (18 headers 16 lines) ---
Using INVITE request as basis request - [email protected]
Sending to 85.181.72.90 : 5060 (NAT)
Found user '00013'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 99
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 85.181.72.90:7078
Found description format G726-32
Found description format G726-32
Found description format G726-40
Found description format G726-24
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x41c (ulaw|alaw|g726|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 00064 in philippschneider (domain voip-server.no-ip.info)
list_route: hop: <sip:[email protected];uniq=DA342FCCC433313B8A3A90FFD5BA1>
Transmitting (NAT) to 85.181.72.90:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.181.72.90:5060;branch=z9hG4bK8BED09BA1E2FE891;received=85.181.72.90
From: <sip:[email protected]>;tag=22F223A07CE0FC39
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 3673 INVITE
User-Agent: FRITZ!Box Fon WLAN 7050 Firmware Version 14.04.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
We're at 217.20.120.238 port 12052
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 85.181.72.90:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.181.72.90:5060;branch=z9hG4bK8BED09BA1E2FE891;received=85.181.72.90
From: <sip:[email protected]>;tag=22F223A07CE0FC39
To: <sip:[email protected]>;tag=as740025fc
Call-ID: [email protected]
CSeq: 3673 INVITE
User-Agent: FRITZ!Box Fon WLAN 7050 Firmware Version 14.04.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 1873 1873 IN IP4 217.20.120.238
s=session
c=IN IP4 217.20.120.238
t=0 0
m=audio 12052 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=silenceSupp:off - - - -

---
vs1128*CLI>
<-- SIP read from 85.181.72.90:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 85.181.72.90:5060;branch=z9hG4bK5FACFFF8C1E82AF3
From: <sip:[email protected]>;tag=22F223A07CE0FC39
To: <sip:[email protected]>;tag=as740025fc
Call-ID: [email protected]
CSeq: 3673 ACK
Contact: <sip:[email protected];uniq=DA342FCCC433313B8A3A90FFD5BA1>
Max-Forwards: 70
User-Agent: FRITZ!Box Fon Eumex300IP 15.04.27 (Sep  6 2006)
Content-Length: 0


--- (10 headers 0 lines) ---
vs1128*CLI>
<-- SIP read from 85.181.72.90:5060:
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 85.181.72.90:5060;branch=z9hG4bK93BE2BB2305BBDA5
From: <sip:[email protected]>;tag=22F223A07CE0FC39
To: <sip:[email protected]>;tag=as740025fc
Call-ID: [email protected]
CSeq: 3674 BYE
Proxy-Authorization: Digest username="00013", realm="asterisk", nonce="35b76022", uri="sip:[email protected]", response="c70acf81f5f84aff5f0784b2efc3f2f7", algorithm=MD5
X-RTP-Stat: PS=200;OS=48000;SP=0/0;SO=0;PR=19;OR=3040;CR=0;SR=0;PL=1;BL=1;EN=PCMU;DE=PCMU;JI=0
Reason: Q.850; cause=16; text="(null)"
Max-Forwards: 70
User-Agent: FRITZ!Box Fon Eumex300IP 15.04.27 (Sep  6 2006)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Content-Length: 0


--- (14 headers 0 lines) ---
Sending to 85.181.72.90 : 5060 (NAT)
Transmitting (NAT) to 85.181.72.90:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.181.72.90:5060;branch=z9hG4bK93BE2BB2305BBDA5;received=85.181.72.90
From: <sip:[email protected]>;tag=22F223A07CE0FC39
To: <sip:[email protected]>;tag=as740025fc
Call-ID: [email protected]
CSeq: 3674 BYE
User-Agent: FRITZ!Box Fon WLAN 7050 Firmware Version 14.04.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
Destroying call '[email protected]'

Die gsm Dateien werden jedoch aufgezeichnet.

Kann mir jemand bitte helfen?

Vielen Dank

Nachtrag: Ich glaube ich habe ein Denkfehler: Mit Record kann nur ein Textaufgenommen werden, der vom Anrufer gesagt wird. Mittels MONITOR kann ich ein Gespräch aufzeichnen.!!

 

[betateilchen]

Nachtrag: Ich glaube ich habe ein Denkfehler: Mit Record kann nur ein Textaufgenommen werden, der vom Anrufer gesagt wird. Mittels MONITOR kann ich ein Gespräch aufzeichnen.!!

So isses :mrgreen:

 

[boelkstoff]

Danke betateilchen ;-)
Mit Monito funktioniert es jetzt.