Der Anrufer hört nix, der Angerufene schon..... ??

REPTILE · Beitrag von **REPTILE** » 13 Feb 2006, 19:36

so habe heute zum ersten mal weng SIPpen probiert...

also ISDN zu ISDN klappt ohne probleme....

habe nun einen SIP Accout als Leitung eingetragen.... hat auch alles geklappt.. ist registriert.... so wenn jetzt jemand Anruft klingelt auch das telefon wie es soll man nimmt ab und dann hört der der Angerufen hat nix von einen aber ich höre den Anrufer... ?!?!

jemand schon mal gehabt ???

Beitrag von **LoUiS** » 13 Feb 2006, 20:17

Hi,

schau mal was fuer Codecs verwendet werden. Ich vermute mal es wurde kein Codec fuer die Rueckrichtung gefunden und evtl. hoert dich der Anrufer deshalb nicht. Wie sind die Codecs eingestellt? Hast Du irgendwelche Codecs ausgeschlossen?

Du kannst auch mal einen "trace # sip call firewall" machen und dabei schauen welche Codecs ausgewaehlt werden.

Ciao
LoUiS

REPTILE · Beitrag von **REPTILE** » 13 Feb 2006, 20:50

hi,

noe codecs sind keine ausgeschlossen, alles angehackt, und auf beste qualität optimiert...

hier mal der trace vom anruf bis zu auflegen vom anrufer..

Code: Alles auswählen

[SIP-Packet] 2006/02/13 20:47:38,330
Receiving Datagram with length 1096 from 80.237.199.17:5060:
INVITE sip:4918051257813176@217.5.232.3:33157 SIP/2.0

Max-Forwards: 10

Record-Route: <sip:80.237.199.17;ftag=as1d133c42;lr=on>

Via: SIP/2.0/UDP 80.237.199.17;branch=z9hG4bK7d65.33c51f16.0

Via: SIP/2.0/UDP 80.237.199.3:5060;branch=z9hG4bK476e3591

From: "+491791478867" <sip:+491791478867@80.237.199.3>;tag=as1d133c42

To: <sip:4918051257813176@siplogin.de>

Contact: <sip:+491791478867@80.237.199.3>

Call-ID: 5faf74f1148d72db29d10ab02a236e69@80.237.199.3

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Mon, 13 Feb 2006 19:47:38 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY

Content-Type: application/sdp

Content-Length: 443



v=0

o=root 19120 19120 IN IP4 80.237.199.3

s=session

c=IN IP4 80.237.199.3

t=0 0

m=audio 15260 RTP/AVP 0 8 97 3 2 18 4 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:97 iLBC/8000

a=rtpmap:3 GSM/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

m=video 14660 RTP/AVP 31 34

a=rtpmap:31 H261/90000

a=rtpmap:34 H263/90000



[Callmanager] 2006/02/13 20:47:38,340
From: +491791478867@80.237.199.3Line: 'RHTEC'
To  : 9106@siplogin.de <Number is complete>
Info: Parse call routing table for active entries
Info: Search in the local user database
Info: User '9106@siplogin.de' not found
Info: Ignore domain
==> From: +491791478867@80.237.199.3Line: 'RHTEC'
    To  : 9106@FPSLine: 'USER.ISDN'
InitiateCall:  9106@FPSLine: 'USER.ISDN'

[Callmanager] 2006/02/13 20:47:38,350
ClnID From: '+491791478867' 
Convert CldID: '9106' ==> CldMSN: '9106'

[SIP-Packet] 2006/02/13 20:47:38,360
Sending Datagram with length 500 to 80.237.199.17:5060:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 80.237.199.17;received=80.237.199.17:5060;branch=z9hG4bK7d65.33c51f16.0

Via: SIP/2.0/UDP 80.237.199.3:5060;branch=z9hG4bK476e3591

From: "+491791478867" <sip:+491791478867@80.237.199.3>;tag=as1d133c42

To: <sip:4918051257813176@siplogin.de>;tag=3307893598-1653946799

Call-ID: 5faf74f1148d72db29d10ab02a236e69@80.237.199.3

CSeq: 102 INVITE

Max-Forwards: 70

User-Agent: LANCOM PBX

Server: jack_uplink_de

Allow: INVITE, ACK, CANCEL, BYE

Content-Length: 0





[SIP-Packet] 2006/02/13 20:47:38,560
Sending Datagram with length 501 to 80.237.199.17:5060:
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 80.237.199.17;received=80.237.199.17:5060;branch=z9hG4bK7d65.33c51f16.0

Via: SIP/2.0/UDP 80.237.199.3:5060;branch=z9hG4bK476e3591

From: "+491791478867" <sip:+491791478867@80.237.199.3>;tag=as1d133c42

To: <sip:4918051257813176@siplogin.de>;tag=3307893598-1653946799

Call-ID: 5faf74f1148d72db29d10ab02a236e69@80.237.199.3

CSeq: 102 INVITE

Max-Forwards: 70

User-Agent: LANCOM PBX

Server: jack_uplink_de

Allow: INVITE, ACK, CANCEL, BYE

Content-Length: 0





[Firewall] 2006/02/13 20:47:46,760
Session opened by rule: internal (SIP)
UDP: 217.5.232.3:48467 => 80.237.199.3:15260
Add Rx minimum of 80 kBit/s to WAN-LINE (result 80 kBit/s)
reduce PMTU on WAN-LINE to 512
Add Tx minimum of 80 kBit/s to WAN-LINE (result 80 kBit/s)


[SIP-Packet] 2006/02/13 20:47:46,760
Sending Datagram with length 716 to 80.237.199.17:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 80.237.199.17;received=80.237.199.17:5060;branch=z9hG4bK7d65.33c51f16.0

Via: SIP/2.0/UDP 80.237.199.3:5060;branch=z9hG4bK476e3591

From: "+491791478867" <sip:+491791478867@80.237.199.3>;tag=as1d133c42

To: <sip:4918051257813176@siplogin.de>;tag=3307893598-1653946799

Call-ID: 5faf74f1148d72db29d10ab02a236e69@80.237.199.3

CSeq: 102 INVITE

Max-Forwards: 70

User-Agent: LANCOM PBX

Server: jack_uplink_de

Allow: INVITE, ACK, CANCEL, BYE

Contact: <sip:4918051257813176@217.5.232.3:33157>

Content-Type: application/sdp

Content-Length: 136



v=0

o=Lancom 12345 6789 IN IP4 217.5.232.3

s=Lancom PBX

c=IN IP4 217.5.232.3

t=0 0

m=audio 48467 RTP/AVP 8 

a=rtpmap:8 PCMA/8000



[SIP-Packet] 2006/02/13 20:47:46,790
Receiving Datagram with length 409 from 80.237.199.3:5060:
ACK sip:4918051257813176@217.5.232.3:33157 SIP/2.0

Via: SIP/2.0/UDP 80.237.199.3:5060;branch=z9hG4bK7371c027

From: "+491791478867" <sip:+491791478867@80.237.199.3>;tag=as1d133c42

To: <sip:4918051257813176@siplogin.de>;tag=3307893598-1653946799

Contact: <sip:+491791478867@80.237.199.3>

Call-ID: 5faf74f1148d72db29d10ab02a236e69@80.237.199.3

CSeq: 102 ACK

User-Agent: Asterisk PBX

Content-Length: 0





[Firewall] 2006/02/13 20:47:46,820
Actions updated by rule: internal (SIP)
remove Rx minimum of 80 kBit/s from WAN-LINE (result 0 kBit/s)
remove Tx minimum of 80 kBit/s from WAN-LINE (result 0 kBit/s)

Add Rx minimum of 80 kBit/s to WAN-LINE (result 80 kBit/s)
Add Tx minimum of 80 kBit/s to WAN-LINE (result 80 kBit/s)


[Firewall] 2006/02/13 20:47:46,870
Actions updated by rule: internal (SIP)
remove Rx minimum of 80 kBit/s from WAN-LINE (result 0 kBit/s)
remove Tx minimum of 80 kBit/s from WAN-LINE (result 0 kBit/s)

Add Rx minimum of 80 kBit/s to WAN-LINE (result 80 kBit/s)
Add Tx minimum of 80 kBit/s to WAN-LINE (result 80 kBit/s)


[SIP-Packet] 2006/02/13 20:48:02,470
Receiving Datagram with length 409 from 80.237.199.3:5060:
BYE sip:4918051257813176@217.5.232.3:33157 SIP/2.0

Via: SIP/2.0/UDP 80.237.199.3:5060;branch=z9hG4bK73258776

From: "+491791478867" <sip:+491791478867@80.237.199.3>;tag=as1d133c42

To: <sip:4918051257813176@siplogin.de>;tag=3307893598-1653946799

Contact: <sip:+491791478867@80.237.199.3>

Call-ID: 5faf74f1148d72db29d10ab02a236e69@80.237.199.3

CSeq: 103 BYE

User-Agent: Asterisk PBX

Content-Length: 0





[Firewall] 2006/02/13 20:48:02,640
remove Rx minimum of 80 kBit/s from WAN-LINE (result 0 kBit/s)
remove Tx minimum of 80 kBit/s from WAN-LINE (result 0 kBit/s)


[SIP-Packet] 2006/02/13 20:48:04,460
Receiving Datagram with length 409 from 80.237.199.3:5060:
BYE sip:4918051257813176@217.5.232.3:33157 SIP/2.0

Via: SIP/2.0/UDP 80.237.199.3:5060;branch=z9hG4bK73258776

From: "+491791478867" <sip:+491791478867@80.237.199.3>;tag=as1d133c42

To: <sip:4918051257813176@siplogin.de>;tag=3307893598-1653946799

Contact: <sip:+491791478867@80.237.199.3>

Call-ID: 5faf74f1148d72db29d10ab02a236e69@80.237.199.3

CSeq: 103 BYE

User-Agent: Asterisk PBX

Content-Length: 0





[SIP-Packet] 2006/02/13 20:48:04,460
Sending Datagram with length 437 to 80.237.199.3:5060:
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 80.237.199.3:5060;received=80.237.199.3:5060;branch=z9hG4bK73258776

From: "+491791478867" <sip:+491791478867@80.237.199.3>;tag=as1d133c42

To: <sip:4918051257813176@siplogin.de>;tag=3307893598-1653946799

Call-ID: 5faf74f1148d72db29d10ab02a236e69@80.237.199.3

CSeq: 103 BYE

Max-Forwards: 70

User-Agent: LANCOM PBX

Server: jack_uplink_de

Allow: INVITE, ACK, CANCEL, BYE

Content-Length: 0

REPTILE · Beitrag von **REPTILE** » 14 Feb 2006, 09:27

so was ich noch getestet habe....

ein LinkSys ATA adapter genommen da den SIP account eingetragen und es geht problemlos!!! auch mal alle Codec bis auf G.711 gesperrt im LinkSys. geht trotzdem, das selbe beim LANCOM gemacht das er eigendlich nur noch den G.711 verwenden kann, das selbe spiel... immer der externe Hört nix.....